Converged Telephony Solution. Technical White Paper

Transcription

1 CTS White Paper Page 1 of 11 Converged Telephony Solution Technical White Paper ٠ May 2004

2 CTS White Paper Page 2 of 11 Converged Telephony Solution White Paper The focus of this white paper is to explain the technology behind the AYCTelecom IPcts and the features & Benefits of using IP and the SIP protocol to converge voice over a data network. This paper is organized into four main sections. IPcts solution overview Application Examples Why Voice over IP? SIP protocol implementation versus H.323 IP Cts Solution overview The AYCTelecom IPcts is an IP telephony solution that uses the SIP (Session Initiation Protocol) standard to provide toll quality voice over a data network. Modular in design, the IPcts is a 1RU high front cabled PBX system capable of supporting the following user configurations, 25 users in a Standard Configuration 100 users in a High Density Configuration 250 users in an Ultra Density Configuration All configurations include, Voic o A 1 hour voice mail system is included within the IPcts chassis which is upgradeable to 8 hours. Automated Attendant o Unlimited level, user configurable attendant system for automatically routing callers to the appropriate user/department using keypad selectable menus. Least cost Routing, Carrier Pre-Selection, Automatic Route Selection o The system will automatically select the cheapest path for an outgoing call, whether the path is through an alternative operator using analogue, ISDN or IP trunks or over a GSM mobile network. Open Licensed Computer Telephony Interface o The CTI is a small program which runs on a users PC if required. If used then it will automatically pop up information regarding the incoming/outgoing voice call such as callers name, address, company etc. The user may also view a phonebook, dial from screen, initiate a call-back and send an

3 CTS White Paper Page 3 of 11 instant message using the CTI. A built in busy lamp field presents the user with an overview of all internal call activity and a simple graphical Divert menu enables the user to manage their active diverts quickly and easily. The CTI is un-licensed and is bundled with the IPcts hardware. 8 SIP trunk lines o These trunk lines enable the IPcts to be connected to a SIP service provider who will be able to switch and terminate Voice over IP traffic throughout their network. Using a SIP service provider follows exactly the same format as using a standard analogue or digital service provider; the only difference is the transport topology for the voice. A number of module cards can be installed in the chassis including, 8 port Analogue Line Card 4 Channel BRI Digital Line Card 8 Channel BRI Digital Line Card 30 Channel PRI Digital Line Card The IPcts is fully user configurable and is simply managed via a web browser. The web based management package bundled with the IPcts hardware enables users with access to the corporate LAN and appropriate login credentials to remotely manage the IPcts. This can be extremely useful if the maintainer of the voice system is either a remote worker, a centralised engineer with responsibilities for multiple buildings/offices or an external maintenance company with dial in access to the Telephone system.

4 CTS White Paper Page 4 of 11 Application Examples Illustrated below are some examples of architecture and topology that would benefit from the installation of voice convergence using an IPcts system. For comparison, included are two versions of each scenario, the first illustration explains how a conventional system would achieve the required features, the other explains how the AYCTelecom IPcts fulfils the solution. Case 1. Cat5 Cabled building with open plan desk configuration Scenario. An update of an existing telephone system is required and the installation requires voic and computer based screen popup s of callers details. We can clearly identify the differences between the two illustrations above. The implementation of the AYCTelecom IPcts has eliminated the requirement for separate structured cabling dedicated for carrying voice traffic. Each IP phone has an in built two port hub, this helps reduce cabling further by passing through IP connectivity from the network, through the phone and onto the computer. This means that only one cable is required per user. In

5 CTS White Paper Page 5 of 11 cases whereby an office has large clustered open plan desk arrangements, only one IP connection will be required to the area and then shared among multiple users by using a network hub/switch. Another shortfall of the first illustration in the scenario above is that a standard analogue PBX does not support CTI to the desktop. The IPcts supports a fully unlicensed CTI package for each Telephone user. Case 2. Company acquires a nearby building and requires cheap voice connectivity between the two buildings. Scenario. A company has acquired a closely located building for warehousing of product and requires a wireless solution for 4 telephone users in the warehouse, 2 wireless laptops, 2 wireless PDA s and 2 static land line phones. Aside from the obvious benefit of reduced cabling which is present in the above case similar to Case 1, the major cost saving benefits of this scenario are: No Separate DECT For VoIP In the first example, the PBX is fitted with an external DECT solution, some vendors have support for DECT handsets within their PBX solutions but these are regarded as add on's and will therefore present the customer with additional charge for the system. Because of this, it is easier to think of DECT as a distinctly separate telephony entity. Using a VoIP solution enables voice traffic to pass over inexpensive wireless data networks without the need for specialist modules/hardware to enable wireless communication. The benefits of such a topology don t stop there however, DECT wireless

6 CTS White Paper Page 6 of 11 base stations offer concurrent telephony connectivity to a low number of wireless devices, that is to say that if four wireless devices required to make a call at the same time then the DECT base station would have to be able to support four concurrent sessions and would need to be hard wired with four extensions from the PBX. This is not the case with Wireless VoIP. One Wireless access point can support up to 32 devices in concurrent communication and requires only one CAT5 cable to connect into the network and back into the IP PBX. Free Intra-Location/Intra-Office communication All of the voice communication from the central office to the warehouse in the first illustration is carried over the PSTN (Public Switched Telephone Network) which is clearly chargeable from the voice operator. When using an AYCTelecom IPcts VoIP solution all voice traffic is passed over the corporate intra-office IP connection between the two sites. This communication is effectively free (aside from the IP connectivity subscription which we assume will need to be in place for the data network in both scenarios). All voice calls are managed centrally and the users within the warehouse can work as though they were sitting at a desk within the central office. All features from the central IP PBX system are available to the warehouse and users can access Caller information either by the LCD panel on the handsets or by a graphical CTI on the warehouse computers if required. Clearly using VoIP to provide intra-office communication in this case negates the need for a separate PBX system at the remote warehouse location. Centralised Billing Managing two locations can be tricky enough without having to administer two separate phone bills. In the first illustration, the voice network at the warehouse location requires a separate PBX and intra-office communication is achieved through the PSTN. Clearly this will generate two operator bills, one for the warehouse and one from the central office. Using VoIP to manage all voice calls centrally over a data network connection between the two locations ensures that there will only ever be one telephone bill generated for all company telephony activity, this is a major advantage for those who are required to administer and assess the corporate phone bill.

7 CTS White Paper Page 7 of 11 Case 3. Remote Workers and Telecommuters Scenario. A company increases it s home based sales team and requires a solution to decrease the call costs between the remote workers and the office, they also require a centralised billing system. The first illustration example above requires that a central VoIP gateway device be connected to the PBX with analogue lines. The number of analogue lines required for connectivity between the PBX and the VoIP gateway will depend upon how many Home workers will need to be on the telephone at the same time. A smaller remote VoIP gateway will also have to be installed at each home workers site to act as the interface between the IP network and an analogue phone. Features that are available to the centralised work force on digital extensions, i.e. CLI(Calling Line ID), Hold/Transfer etc. may not be available to the home workers as the topology is essentially a solution which simply encapsulates and transports legacy analogue technology. Using the AYCTelecom IPcts, which is a true VoIP PBX, eliminates the previous problems. The number of concurrently supported home worker conversations into the corporate centralised IP PBX depends solely upon the availability and bandwidth of the company s wide area network connection. When you consider that the average SME business connectivity to the internet is 2Mb, and each IP voice call compressed using the G.729 codec uses just 8KB/s bandwidth, the number of concurrent remote sessions should not present any particular bandwidth problems. In fact, using these figures, and

8 CTS White Paper Page 8 of 11 including IP overhead, we can calculate that a 2Mb/s connection can support up to 240 G.729 voice calls, or 240 home workers. The IPcts performs QOS on it s dedicated WAN port to ensure that critical voice traffic is transported at the highest possible priority over bursty internet traffic. You should also notice that by using a true VoIP PBX system, the requirement for both centralised and remote VoIP gateway units is eliminated, therefore cutting hardware cost dramatically. All voice calls and PBX features are passed over the IP network and each home worker can access these PBX features as though they are located within the central office. If the home worker is running a CTI session on their computer then they will be able to view a busy lamp field which shows the status of all other extensions on the PBX system, access the centralised contacts database, and will be able to send instant messages to colleagues via the on-screen display.

9 CTS White Paper Page 9 of 11 Why Voice Over IP? A question that historically could simply be answered with financial reasoning, however hardware vendors are experiencing a serious upsurge in the interest of small to medium sized telephony solutions that include the types of features found only in larger enterprise communication solutions. The nature of VoIP s single converged path for both voice and data signalling enables these extended features to be included in smaller telephony solutions which are more appropriately sized and priced for the small to medium size enterprise. Computer screen pop ups presenting the user with information regarding the callers name/number/address, tailored automated attendant menu s and digital music on hold are just some of the features that are a requirement for small to medium enterprise businesses, and VoIP technology can deliver these features in a non-complex cost effective manner. Clearly the interest in VoIP isn t driven solely by the features that can be offered, cost benefits are always a major factor in the procurement of communication systems and VoIP offers many ways to reduce both the implementation cost and the ongoing operational costs of a communications network. As we saw in the application example cases above, the first cost benefit at installation phase of a VoIP system is the decrease in the amount of cabling required for the office/location. The SIP IP phones include an integral mini two-port hub enabling an Ethernet cable loop through from the phone out to a PC. This ensures that cabling is kept to a minimum and that one user will only require one cable to their desktop. One of the major cost benefits of an IP based telephony system is in the case of either remote teleworking or intra office communication. Taking the teleworker as a first example; the teleworker would have a broadband internet connection and VPN connectivity into the central office (see Case. 3 above). The teleworker would have an IP phone attached to their local network which would gain connectivity to the central IPcts telephone system using the corporate VPN. The teleworker could then make and receive calls as though they were sitting in the office. Optionally the teleworker could boot up their computer and run a CTI window in order to view a busy lamp field containing information about the status of other users phones on the system. All of this functionality is provided with no associated call cost because the entire voice and data systems run over the broadband connection. Similarly, if we analyse the case of intra office communication, assuming that IP connectivity exists between the two locations, all intra office voice communication can be passed across the IP network at no additional cost. Optionally, one site can be designated as the HeadQuarters and all external voice communication can be managed and billed centrally at the HQ. (See Case. 3) In the past it has been an uphill struggle to convince the data community to accept voice overlaid onto the network. However, with compression algorithms as good as they are currently and the average internal bandwidth available per user, the argument against VoIP on a corporate LAN has effectively been extinguished. The IPcts supports SIP phones which use a

10 CTS White Paper Page 10 of 11 compression algorithm to downsize each voice packet and use just 8Kb/s data bandwidth. When you perform the math on this figure we can work out that for the duration of a call, one voice call will take just 0.008% bandwidth of a 100MbT network. SIP protocol implementation versus H.323 The AYCTelecom IPcts adheres stringently to the SIP (Session Initiation Protocol) standard. This enables the system to support all SIP based IP phones. As the name suggests, the SIP protocol is used to initiate a communication session between two devices, once the session has been connected, the initiator (in this case the AYCTelecom IPcts) is no longer part of the conversation and the two end devices talk over the network independently. The H.323 protocol works in a different way, it performs the handshaking and negotiation of two end points and connects them together, however the H.323 server will stay in the loop and manage the communication for the entire duration of the call. The differences in the protocols is largely due to the applications that they were designed for. H.323 was primarily designed for the connection of multimedia codecs across IP networks and as such includes specific targeted support for a large number of multimedia transport mechanisms. This, coupled with the fact that it s call setup signalling is ISDN style Q.931 and it implements RAS and H.245 for terminal/gatekeeper negotiations and codec/facilities handshaking, make it a very heavyweight protocol with aspects of the stack unused and redundant in a voice only application. SIP was designed to be a more flexible protocol that simply initiates the requested connection and hands off all higher level processes to the communicating end points. This makes SIP a very lightweight, and therefore processor light, protocol in comparison. The major benefit of using the SIP protocol is that once the server has initiated a call, it goes back into wait/dormant state and waits for the next SIP request from the network. While it is in this wait state, the processor and memory are simply running background processes and are not being throttled by call management. This means that much less processing power and memory is needed for call switching and can be assigned to other processes such as embedded voic , automated attendant etc. With processing power and memory requirements at a minimum due to the implementation of SIP, the AYCTelecom IPcts can be offered to the market place at very competitive prices compared to H.323 based devices. Conclusion In this document we have talked about various technologies, topologies and architectures that will ultimately enable businesses world wide to converge their voice and data networks in order to streamline their telecommunication operations and therefore reduce the cost of running a business. We have investigated specific application scenario s and proven that Voice Over IP is a realistic, efficient and cost effective way to achieve cost effective company

11 CTS White Paper Page 11 of 11 communication both internally and externally to an organization. In reality, we have all used VoIP at some point, with the plethora of backend international carrier networks running their voice over IP networks it s actually pretty hard for an internationally trading company to avoid using the technology on a day to day basis without realising. The battle between the heavy weight, application specific H.323 protocol versus the relatively new light weight, flexible SIP protocol solution will no doubt rage on with points for and against both protocols in certain applications. The general opinion, for the specific application of Voice over IP, is that the SIP standard with it s text based protocol messages and tiny processor load, will become the most popular protocol for the development and deployment of reduced cost Voice over IP equipment. The protocols small amount of negotiation message packets and lack of call management beyond call initiation will be very attractive to the data world and IT managers wanting to keep bandwidth hit to a minimum. To summarise, Voice over IP using the SIP standard will be the technological direction in which small to medium sized enterprise companies will move in order to converge voice and data and make huge savings on voice communication. AYCTelecom have the solution to make this financial dream a reality.