Software Requirements Specification <VoIP SOFT PBX > Project Code: SPBX Internal Advisor : Aftab Alam Associate professor FAST NU Lahore Pakistan External Advisor: Asad Gill TRG pakistan Project Manager: Wajahat Iqbal Project Team: Umair Ashraf Imran Bashir Khadija Akram Submission Date: 23rd November, 2007 Project Manager s Signature
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Document Information Category Customer Project Document Document Version Identifier Status Author(s) Approver(s) Information FAST-NU VoIP soft PBX Requirement Specifications 1.0 SPBX Draft Umair Ashraf,Khadija,Imran Bashir Mr Aftab Alam Issue Date 23rd November 07 Document Location Supervisor : Mr Aftab Alam Distribution External Advisor: Mr Asad Akram PM :Mr Wajahat Iqbal Page 3 of
Definition of Terms, Acronyms and Abbreviations Term RS PBX SIP RTP VoIP IETF Description Requirements Specifications Private Branch Exchange Session Initiation Protocol Real Time Protocol Voice over IP Internet engineering Task Force Page 4 of
Table of Contents 1. Introduction... 6 Purpose of Document... 6 Project Overview... 6 Abstract 6 Introduction... 7 Why should we use VOIP?... 7 Lower Equipment Cost... 8 Widespread availability of IP... 9 Vision and Scope... 9 The Virtual PBX (Private Branch Exchange)... 9 2. Functional Requirements... 10 Registering a user... 10 Dialing and placing Call... 10 Accepting (call pick up) /rejecting a call... 10 Terminating the Session... 10 Voice Conferencing... 11 Missed Call Alert... 11 Call Hold. 11 Call Detail Reporting. (CDR)... 11 Caller identification... 12 Dial Plan 12 Speed dialing... 12 Administrative and management services... 12 Phone book service... 12 Call Recording... 13 Voice mail 13 Call Transfer... 13 Call forwarding... 13 3. Non-functional Requirements... 14 Performance and Reliability... 14 Robustness... 14 Security 15 Standards compliance... 15 Usability 15 Portability 15 Development tools... 15 Documentations... 16 Maintenance and Support... 16 Hardware Requirements... 16 4. References... 16 5. Appendices... 17 Real-time Transport Protocol (RTP)... 17 Session Initiation Protocol (SIP)... 18 Jitter G.729 Codec Page 5 of
1. Introduction Purpose of Document The Purpose of this Document is to define the Scope and boundary of the System to be developed. The basic architecture and the functionalities of the system will be built on the basic of these requirement specifications. Each Requirement specifies the functionality to be expected from the system. This document is for the all the stakeholders I-e the Developers, Users and the client of the product like university admission department.. Project Overview Abstract VoIP is the innovative and rapidly growing technology that is being rapidly adopted all over the world by the companies in the Communications and IT industry. Although voice over IP (VoIP) has been in existence for many years, it has only recently begun to take off as a viable alternative to traditional public switched telephone networks (PSTN). Interest and acceptance has been driven by the attractive cost efficiencies that organizations can achieve by leveraging a single IP network to support both data and voice. But cost is not enough to complete the evolution; service and feature parity is a main requirement. Customers will not accept voice quality or services that are less than what they are used to with a PSTN and, until now, VoIP fell short in Delivery. Page 6 of
Introduction VoIP is simply the transport of voice traffic by using the Internet Protocol (IP). It is a technology that allows you to make voice calls over a broadband internet connection instead of a regular phone line. In VOIP your voice is converted into a digital signal that travels over the internet. If you are calling a regular phone number (i.e. on a PSTN), the signal is converted to a regular telephone signal. Why should we use VOIP? Traditional telephony carriers use circuit switching for carrying voice traffic. Circuit switching was designed for voice from the outset; hence it carries voice in an efficient manner. However it is an expensive solution. Nowadays people want to talk much more on phone, but they also want to communicate in a myriad of other ways through e-mail, instant messaging, video, the World Wide Web, etc. Circuit switching is not suitable for this new world of multimedia communication. IP is an attractive choice for voice transport for many reasons, including the following:- Lower equipment cost Integration of voice and data applications Lower bandwidth requirements Widespread availability of IP Page 7 of
Lower Equipment Cost The IP world is different from the monolithic systems of mainframe computers and circuit-switching technology. IP systems tend to use distributed client-server architecture rather than large monolithic systems, which means that starting small and growing as demand dictates is easier. IP architectures and standards are more open and flexible plus they are competition friendly, than telephony standards, enabling the implementation of unique features so that a provider can offer new features more quickly. Hence, the range of choices is large; the equipment cost is drastically lower than that of circuit switching products; and the pace of development is incredibly fast. Voice/Data Integration and Advanced Services IP is the standard for data transactions- everything from e-mail to web browsing to e-commerce. When we combine these capabilities with voice transport on a single network, we can easily imagine advanced features that can be based on the integration of the two. Lower Bandwidth Requirements Circuit-Switched telephone networks transport voice at a rate of 64 Kbps. Typical human speech has a bandwidth of 4000 Hz, and according to the SyQuest theorem, when we would digitize this voice a telephone system would take 8000 samples per second. In VoIP we can use coding schemes which enable speech to be transported between 6 Kbps to 32 Kbps. Therefore, VoIP offers significant advantages over circuit-switching from a bandwidth point of view. However we would be using G.711 for this purpose which is also at a rate of 64 Kbps. Page 8 of
Widespread availability of IP IP is practically everywhere. Every personal computer produced today supports IP. IP is used in corporate LANs and WANs, dial-up internet access etc. As a result there is widespread availability of IP expertise and numerous application-development companies. This factor alone makes IP a suitable choice for transporting voice. Vision and Scope The Virtual PBX (Private Branch Exchange) Numerous IP based PBX solutions are in place and more are being deployed daily. The idea of an IP-based PBX is useful. Firstly, this system integrates the corporate telephone system with the corporate computer network, removing the need for two separate networks. A new office is wired for voice communication, as well. The PBX itself becomes just another server or group of servers in the corporate LAN, which helps to facilitate voice/data integration. We would be implementing a soft PBX that is going to be a server that performs call-routing functions, replacing the traditional legacy PBX or key system. Our PBX would allow a number of attached soft phones to make calls to one another and to connect to other telephone services. The basic software would include many features available in proprietary PBX systems: voice mail, conference calling, interactive voice response, and automatic call distribution, just to name a few. Our system would also help the user in building the dial plan for the network. Page 9 of
We would be using the Session Initiation Protocol (see appendix 1) as our VoIP protocol. Our PBX would be acting as both the registrar and as a gateway between the soft phones. 2. Functional Requirements Registering a user The system shall allow a user to register itself to the PBX by its IP Address and caller ID through a soft phone. Dialing and placing Call The system shall allow users to dial and place a call to each other using Soft phones through a PBX over the Network (internet or intranet) Accepting (call pick up) /rejecting a call The system shall allow the user to receive or reject a call of the caller using Soft Phone. Terminating the Session The system shall allow the user to terminate the call at any time. Page 10 of
Voice Conferencing The System shall allow users to have voice conferencing service amongst multiple users at a time. Missed Call Alert The System shall support missed call service and provide the information of the missed calls to the user. Call Hold. The system shall provide the station user, to hold a call in progress, the ability to dial either an appropriate hold code, or to depress a feature button which would place the call in progress on system hold, and allow the station user to: (1) originate another call or, (2) use any other features provided by the system. Call Detail Reporting. (CDR) The PBX shall be equipped to capture Call Detail information. The information to be captured shall as a minimum provide: o Date of Call (Month and Day) o Calling Station Number o Called Number (all depressed digits) o Time (Time Call Was Placed) o Duration of Call (Minutes and Seconds) Page 11 of
Caller identification The System shall allow the user to get the information of the caller person like his/her caller id and the name. Dial Plan The System shall provide complete dial plan option to the users. Speed dialing The system shall support the services of PBX. the speed dialing through a Administrative and management services The system shall provide the administrative and management services of the PBX like changing dial plans, managing users, managing passwords and creating call Detail Reporting services. Phone book service The system shall support and provide phone book service to the users. Page 12 of
Call Recording The system shall allow the user to record the call of the conversation. Voice mail The system shall support and provide voice mail facility to the users. Call Transfer The system shall allow the user to transfer his/her call to another user at a different location. Call forwarding The System shall allow the user to forward his/her call to another location. Page 13 of
3. Non-functional Requirements Performance and Reliability System must be scalable up to 50 devices per PBX. The system must ensure that sending and receiving packets are not discarded, and a mechanism must be adopted so that packet loss and retransmission should not occur due to algorithm used in the application otherwise, voice quality or service disruptions might occur. The jitter buffer (see appendix for details) configuration must be implemented in the software to avoid packet delay which should not be more than 100 millisecond between the two consecutive packet transmissions otherwise quality of voice may drop. The system must ensure to utilize as low bandwidth of the network as possible.inorder to prevent high traffic over the internet by the application some compression codec s like G.729 (see appendix for more details) shall be supported by the system. Robustness The system should support proper exception handling like incase of unavailability of Network. Page 14 of
Security The system shall provide complete security and privacy to the users and no other party shall be allowed to listen to the conversation between the two end users. Standards compliance The system shall fulfill all the standards of the IEEE and IETF. The system shall support all standard protocols like SIP and RTP protocols. (See Appendix for details). Usability The system must be providing user-friendly interface to the end users. A dial pad and telephone like features must be provided with the interface so as to provide the end user a Complete Soft phone on his/her desktop. The system shall provide an interface to the Administrator to maintain and configure the system. Portability The system shall work under Microsoft Windows XP or Microsoft Windows 98 Development tools The system shall be developed in rapidly growing and cutting edge technology of.net and C sharp framework. Page 15 of
Documentations The Specification document and user manual shall be provided when the software will be handed over to the client. Maintenance and Support The installation and configuration, maintenance support shall be provided. Hardware Requirements Pentium 4 system with at least 512 Mb of Ram Full duplex Sound Card A Conventional LAN based network or Intranet. 100 Mbytes of Disk space 4. References Carrier Grade Voice over IP by Daniel Collins Understanding VOIP networks by Juniper Network Real-time protocol RFC SIP RFC Voice over IP Fundamentals by Jonathan Davidson How stuff works platform Wekipedia.org Asterisk the Open Source Telephony Platform www.asterisk.org Page 16 of
5. Appendices Real-time Transport Protocol (RTP) Real-time Transport Protocol (RTP), which provides end-to-end delivery services for data with real-time characteristics, such as interactive Audio and video. Services include payload type identification, sequence numbering, time stamping and delivery monitoring. The media gateways that digitize voice use the RTP protocol to deliver the voice (bearer) traffic. The RTP protocol provides features for real-time applications, with the ability to reconstruct timing, loss detection, security, content delivery and identification of encoding schemes. For each participant, a particular pair of destination IP addresses defines the session between the two endpoints, which translate into a single RTP session for each phone call in progress. RTP is an application service built on UDP, so it is connectionless, with best-effort delivery. Although RTP is connectionless, it does have a sequencing system that allows for the detection of missing packets. As part of its specification, the RTP Payload Type field includes the encoding scheme that the media gateway uses to digitize the voice content. This field identifies the RTP payload format and determines its interpretation by the CODEC in the media gateway. A profile specifies a default static mapping of payload type codes to payload formats. With the different types of encoding schemes and packet creation rates, RTP packets can vary in size and interval. Administrators must take RTP parameters into account when planning voice services. All the combined parameters of the RTP sessions dictate how much bandwidth is consumed by the voice bearer traffic. RTP traffic that carries voice traffic is the single greatest contributor to the VoIP network load. Page 17 of
Session Initiation Protocol (SIP) The Session Initiation Protocol is part of IETF's multimedia data and control protocol framework. SIP is a powerful client-server signaling protocol used in VoIP networks. SIP handles the setup and tears down of multimedia sessions between speakers; these sessions can include multimedia conferences, telephone calls, and multimedia distribution. SIP is a text-based signaling protocol transported over either TCP or UDP, and is designed to be lightweight. It inherited some design philosophy and architecture from the Hypertext Transfer Protocol (HTTP) and Simple Mail Transfer Protocol (SMTP) to ensure its simplicity, efficiency and extensibility. SIP uses invitations to create Session Description Protocol (SDP) messages to carry out capability exchange and to setup call control channel use. These invitations allow participants to agree on a set of compatible media types. SIP supports user mobility by proxying and redirecting requests to the user's current location. Users can inform the server of their current location (IP address or URL) by sending a registration message to a registrar. This function is powerful and often needed for a highly mobile voice user base. The SIP client-server application has two modes of operation; SIP clients can ether signal through a proxy or redirect server. Using proxy mode, SIP clients send requests to the proxy and the proxy either handles requests or forwards them on to other SIP servers. Proxy servers can insulate and hide SIP users by proxying the signaling messages; to the other users on the VoIP network, the signaling invitations look as if they are coming from the proxy SIP server. Page 18 of
Jitter Jitter is the measure of time between when a packet is expected to arrive to when it actually arrives. In other words, with a constant packet transmission rate of every 20 ms, every packet would be expected to arrive at the destination exactly every 20 ms. The greatest culprit of jitter is queuing variations caused by dynamic changes in network traffic loads. Another cause is packets that might sometimes take a different equal-cost link that is not physically (or electrically) the same length as the other links. Media gateways have play-out buffers that buffer a packet stream, so that the reconstructed voice waveform is not affected by packet jitter. Play-out buffers can minimize the effects of jitter, but cannot eliminate severe jitter. Although some amount of jitter is to be expected, severe jitter can cause voice quality issues because the media gateway might discard packets arriving out of order. In this condition, the media gateway could starve its play-out buffer and cause gaps in the reconstructed waveform. G.729 Codec G.729 is an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds. G.729 is mostly used in Voice over IP (VoIP) applications like SIP phones for its low bandwidth requirement. Standard G.729 operates at 8 kbit/s, but there are extensions, which provide also 6.4 kbit/s and 11.8 kbit/s rates for marginally worse and better speech quality respectively. It also requires less computation during encoding and decoding. Page of