2011                       Design and Implementation of                       a Phone Card Company                       I...
Table of Contents1.       CHAPTER ONE: INTRODUCTION .........................................................................
4.1.      BILLING SERVER: JERASOFT VOIP CARRIER SUITE (VCS) .................................................................
1. Chapter one: Introduction    1.1. Project IntroductionOne of the main advantages of internet is carrying data as well a...
1.2. InfrastructureFirst, to establish our prepaid calling card system, a solid infrastructure is developed. Clearly, lack...
   Once the call is terminated, the corresponding Call Detail Record (CDR) is saved in the system to        issue stateme...
2. Chapter two: System Requirements,                      Implementation Steps and Protocols    2.1. Requirements and Impl...
2.1.5.      Finding VoIP Termination/Long Distance ProvidersVoIP Termination Providers sell us traffic to various destinat...
Terminals are used for real-time bi-directional multimedia communications, and can be an IP-phone or apersonal computer IP...
Figure 2.1: Class for a H.323 architecture [18]The H.323 applies RTP/RTCP (real-time transport protocol) as its transport ...
2.2.2.1.    SIP ArchitectureThe primary of SIP systems are user agents and servers. User Agents (UAs) are a combination of...
Figure 2.2: SIP Architecture [18]SIP requests and responses include acute information about the satisfied and designs of c...
2.2.3.      Media Gateway Control ProtocolMGCP is a protocol which is produced by the Media Control Working Group and is u...
One of the advantages of IAX is minimize bandwidth for using media transmission. “with specificattention drawn to control ...
3. Chapter Three: System TopologyBecause of high automation and since internet is the main component and other functions c...
On next chapter of the report we will discuss more specifically about Server and SoftswitchConfigurations. Also, you can f...
Call Authorization:       Available BW       Available gateway       IP address       Available money in the accountBi...
have propelled the growth of those services. They are especially popular between mobile phone usersas an alternative to th...
The Cisco VIA solution offers so many profits, such as lower transportation and operating costs comparewith other industry...
significant cost factor in delivering the service, is included in Cisco AS5000 universal gateways. Thisintegrated feature ...
Figure 3.3: Business Voice Services Description [19]The flexibility of the Cisco SP Business Voice Solution architecture w...
means with Cisco technology, service providers can deploy a robust H.323 network and still support SIPtraffic terminating ...
3.4.5.      Wireless Transit SolutionThe wireless transit solution is a slight variant on the wireline transit solution. W...
4. Chapter Four: Billing Server and Softswich    4.1. Billing Server: Jerasoft VoIP Carrier Suite (VCS)In whole sale solut...
Figure 4.1: Jerasoft system GUI overview    Like other vendors BillBery solution supports postpaid and prepaid billing ser...
• Retail subscriber access to feature servers• Retail SIP trunking applications• Wholesale peering applications• Combinati...
on one Virtual IP address and depart on another. This allows consolidation of network        interfaces to be accomplished...
• Centralizes CDRs• Intelligent route hunting• Real time performance monitoring• Protects all internal IP addresses• Reduc...
• License Sharing Zones providing real time sharing of network resources, lowering capital expenses andoperating expenses....
• Ability to have IP PBXs register directly to the VSXi for small business applications, with changeable IPaddresses• Can ...
• SIP, TCP/TLS calls are internetworked to normal SIP or H.323 routesThe VSXi allows carrier interconnects which require d...
• 2U servero 8000 active calls (H.323 or SIP with media) per chassiso Redundancy on fans, power supplies and RAID driveso ...
4.5. VSXi ConfigurationThe VSXi features a Graphic User Interface. All configurations can be accomplished using this GUI.O...
Figure 4.7: Sansay GUI overview4.5.1.    Sansay GUI Major Tab: System (Figure 4.8)                                       ...
o   Basic Tab (Editing Network Settings)Click the system tab to begin to edit the Network Settings. Click the Edit Network...
Edit System PageClick this tab to edit system name, CDR information, payload port, SIP port, H.323 information, NTPinforma...
Current Time, Date, and Time ZoneSet the current time, date and select the time zone for your area. Don’t forget to select...
Figure 4.11: SNMP Tab    o   Advanced TabFrom this tab, you can save your configuration and restore it at a later time. Th...
Reboot SystemReboot System will allow you to reboot the system. This will terminate all current calls. You will begiven an...
Service Ports combined with Routes and Resources, provide intelligent connections between class 5VoIP switches and VoIP te...
Figure 4.14: Service Port AddIndexService Port Index is a unique number to identify your Service Port. You may choose any ...
Port TypePort Type choices are UDP, TCP, TLS, or DTLS. Again this must match the devices you connect to withthis port.Reso...
Figure 4.16: Virtual IP add     Major Tab: Application Servers    o   Radius Servers Tab Radius Servers are principally u...
Figure 4.17: Radius Servers                                         Figure 4.18: Radius GroupGroup IndexEnter a Group Inde...
Group PolicySelect a Group Policy from the drop down list. Choices are round_robin or top down. This refers to theway the ...
Figure 4.20: CNAM Group    o   Local Number Portability Servers (LNP) TabLPN is used to check the dialed number to see if ...
 Major Tab: ResourcesA Resource is any device that will send or receive calls to/from the VSXi. It could be a small IP ga...
Figure 4.22: Adding ResourcesResource TypeSelect the resource type, from the drop down, Peering, Access, or Dynamic Peerin...
will use the configured protocol. However, inbound calls are accepted with SIP GW, SIP Proxy, H.323GW, H.323 GK, or ENUM S...
second basis. CPS limiting measures the CPS (inbound and outbound) on the TID and rejects calls thatexceed that value. Any...
Note: Only trace 1 resource or route at a time. This will ensure that the call being traced will be fromthe desired resour...
Figure 4.23: Registrar    o   Resource Block List TabUnder this tab, you can input digit patterns you wish to block with r...
Figure 4.24: SIP Profile     Major Tab: RoutesA route is designated by an Alias, Digit Match, and the Route Table it appl...
Figure 4.25: RoutesRoutes Add                                      53             Figure 4.26: Route add                  ...
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
Design And Implementation Of A Phone Card Company
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Design And Implementation Of A Phone Card Company

  1. 1. 2011 Design and Implementation of a Phone Card Company Instructor: Dr.He Students: Amiraslan Aslanian Ramin Etezazian Farzaneh MounesanCopyright ©Gathered and writtenbyAmiraslan AslaninanRamin EtezazianFarzaneh Mounesan– Ryerson University Ryerson University2011 8/12/2011
  2. 2. Table of Contents1. CHAPTER ONE: INTRODUCTION ..................................................................................................................... 31.1. PROJECT INTRODUCTION .......................................................................................................................... 31.2. INFRASTRUCTURE ..................................................................................................................................... 4 1.2.1. INTERNET SERVICES .......................................................................................................................................5 1.2.2. PHONE SERVICES ..........................................................................................................................................5 1.2.3. POWER SERVICES ..........................................................................................................................................52. CHAPTER TWO: SYSTEM REQUIREMENTS, IMPLEMENTATION STEPS AND PROTOCOLS ................................ 62.1. REQUIREMENTS AND IMPLEMENTATION STEPS ........................................................................................ 6 2.1.1. PRICE .........................................................................................................................................................6 2.1.2. QUALITY .....................................................................................................................................................6 2.1.3. RELIABILITY ..................................................................................................................................................6 2.1.4. CAPACITY OF SERVICE ....................................................................................................................................6 2.1.5. FINDING VOIP TERMINATION/LONG DISTANCE PROVIDERS ..................................................................................7 2.1.6. FINDING ACCESS NUMBER/DID PROVIDERS .......................................................................................................72.2. VOIP PROTOCOLS ...................................................................................................................................... 7 2.2.1. H.323 ........................................................................................................................................................7 2.2.1.1. H.323 Security......................................................................................................................................8 2.2.2. THE SESSION INITIATION PROTOCOL (SIP) .........................................................................................................9 2.2.2.1. SIP Architecture .................................................................................................................................10 2.2.2.2. SIP Security ........................................................................................................................................10 2.2.2.3. SIP Services ........................................................................................................................................11 2.2.3. MEDIA GATEWAY CONTROL PROTOCOL ..........................................................................................................12 2.2.4. INTER-ASTERISK EXCHANGE ..........................................................................................................................12 2.2.5. PRIMARY RATE INTERFACE AND BASIC RATE INTERFACE......................................................................................133. CHAPTER THREE: SYSTEM TOPOLOGY.......................................................................................................... 143.1. HARDWARE AND DEVICES ....................................................................................................................... 143.2. SOFTWARE AND APPLICATION ................................................................................................................ 153.3. PREPAID AND POSTPAID CALLING CARDS SERVICES ................................................................................ 16 3.3.1. PREPAID AND POSTPAID SERVICE EXPLANATION.................................................................................................183.4. SOLUTIONS.............................................................................................................................................. 18 3.4.1. ADVANTAGE OF CISCO VIA SOLUTION IN CALLING CARDS SERVICES......................................................................18 3.4.2. CISCO SP VOICE SOLUTION ...........................................................................................................................19 3.4.3. GLOBAL LONG DISTANCE SOLUTION ...............................................................................................................20 3.4.4. INTEGRATE TRANSPORT SOLUTION .................................................................................................................21 3.4.5. WIRELESS TRANSIT SOLUTION .......................................................................................................................22 1 FIGURE 3.5: WIRELESS TRANSIT SOLUTION INFRASTRUCTURE [2] ..........................................................................................22 Page4. CHAPTER FOUR: BILLING SERVER AND SOFTSWICH ..................................................................................... 23
  3. 3. 4.1. BILLING SERVER: JERASOFT VOIP CARRIER SUITE (VCS) ........................................................................... 234.2. SOFTSWITCH: SANSAY, THE VSX-INTEGRATED MULTIMEDIA SUBSYSTEM ............................................... 24 4.2.1. VSXI PRIMARY FEATURES .............................................................................................................................254.3. SUBSCRIBER ACCESS APPLICATION .......................................................................................................... 27 4.3.1. SIP TRUNKING APPLICATION .........................................................................................................................28 4.3.2. VSXI WHOLESALE APPLICATION ....................................................................................................................294.4. VSXI HARDWARE ..................................................................................................................................... 304.5. VSXI CONFIGURATION ............................................................................................................................. 32 4.5.1. SANSAY GUI ..............................................................................................................................................335. CHAPTER FIVE: SUMMARY .......................................................................................................................... 636. REFERENCES ................................................................................................................................................ 647. APPENDIX: CONFIGURATIONS ON ROUTER AND SWITCH ............................................................................ 657.1. CONFIGURATION ON SWITCH CISCO 3750G ............................................................................................ 657.2. CONFIGURATION ON ROUTER CISCO 3845 .............................................................................................. 71 2 Page
  4. 4. 1. Chapter one: Introduction 1.1. Project IntroductionOne of the main advantages of internet is carrying data as well as voice, and base on voice over IPtechnology these days no one worries about the cell phone bill since long distance call are offered inreally low rates everyday by companies, everybody gets calling card and uses it for cost-effectiveconversation and saving money.Where is this calling card coming from?Basically whenever there is an internet, we can use this feature to run voice over internet, when youhear about the company you think it’s like huge automobile company while it’s not like that. It’s possiblewith just two devices and a few people to run it and of course some basic knowledge! For having thiscompany we can either have our own equipments specifically for this matter or use the share ISP, whichis better and less maintenance and employee are required. So as a whole sale service provider we rent aline from ISP (not to mention that this lines connectivity should have redundancy and always beguaranteed) and sell the minute’s base on reliability, speed and quality of the lines to retail customers.the device is necessary between source and destination to manage the call, find the best and cheapestpath and finally calculate the price and issue the invoice for retails, in retail part the whole minuteswhich are given by whole sale are broken by small companies and are sold to people in cards; so if weuse ISP as a infrastructure we should just stand in the middle and do this job. All these tasks are done byplatform called Softswitch which we are going to explain all details and functions as well as some view ofVoIP protocols that are being used for this purposes. Here deep knowledge of VoIP structure andconfiguration is not a must, but base on your need and work load the specific vendor and platformwhich fulfills our needs is chosen, and you should just go through the details of that specific Softswitchor software.Here we are going to cover the details about this job based on “Keyhan Telecom Company” and specificdevices and vendors which are being used in its structure.Now you can just sit in your home and run a company! 3 Page
  5. 5. 1.2. InfrastructureFirst, to establish our prepaid calling card system, a solid infrastructure is developed. Clearly, lacking asolid infrastructure will result in wasting time worrying about the system’s integrity, a time that could bepossibly devoted to market the product. We will provide everything including a reliable hosted serverwith high bandwidth, power services and a Calling Card Platform to administrate the business. Figure 1.1presents the list of equipment and services that are required to start the business. Figure 1.1: Required equipment to start calling business [6]The following describes the process shown in the above topology:  The customer buys our calling card online or from a retail store.  Next, the customer dials the local or 1-800 access numbers (DID) shown on the topology using a phone (e.g., a cell Phone, a landline or a payphone).  The Calling Card Platform will respond with an IVR message asking the customer to enter the PIN. If the card is Pin-less the message will prompt the customer to enter the destination number.  Once the PIN is keyed in, the Calling Card Platform will validate the customer, announce the balance of the customer’s account and prompt him/her to dial the destination number.  Next, the amount of available time for the destination is announced by the IVR system.  The Calling Card Platform transports the call over internet to the chosen carrier. 4  Page VoIP provider routes the call to the customer’s intended destination.
  6. 6.  Once the call is terminated, the corresponding Call Detail Record (CDR) is saved in the system to issue statements, create reports, and analyze the business. 1.2.1. Internet ServicesInternet service is the foundation of a calling card network. A reliable Internet service will enable us toreliably keep the billing and VoIP servers online. One should place the system in a co-location facility.Also, one should ask the internet provider about the existing redundancy in case their servicemalfunctions. The provider will often take advantage of another internet provider in this case. Morespecifically, if an internet provider lacks a failover plan, one may consider a second internet service asthe backup. 1.2.2. Phone ServicesChoosing the tight phone service is as important as selecting a reliable internet services. If our phonelines are not working well, the customers are unable to call into the system. Often, phone services aremore reliable compared to Internet services. Thus, it is unnecessary to be as concerned aboutredundancy as internet services. However, we must ensure that the correct phone service is chosen forour application. For example, if our system supports T1 lines, one should check that the T1 is a PRI. 1.2.3. Power ServicesIn most places power outage is common, or perhaps a daily event. If the systems are located in a placewhere power outages are, common a power generator is crucial. Moreover, regardless of the location,one must have a UPS battery backup that lasts at least two hours in case of a power outage. Even if theequipment is placed in a co-location facility, it is still very important to be equipped with UPS batterybackup. 5 Page
  7. 7. 2. Chapter two: System Requirements, Implementation Steps and Protocols 2.1. Requirements and Implementation Steps 2.1.1. PriceIt is common to choose the long distance provider simply based on the price (i.e., the lowest longdistance rate). One factor that is important is the fact that some services have a monthly fee. You maystill be better off with a long distance provider that charges a monthly fee if the rate is quite a bit lower,or if it is slightly lower and you use enough minutes to offset the difference. Another critical factor iswhere your customers call the most. Some long distance services have discounted rates for in-statecalls. Some long distance providers have lower rates for state-to-state long distance. Another importantfactor is the billing increment. If a service is billed in six second increments and you make a lot of shortcalls, it may be a better deal than another long distance provider who offers 60 second increments at aslightly lower long distance rate. Generally, wholesale billing increments should not be greater than 30seconds. 2.1.2. QualityLong distance companies in the VoIP market differ substantially in terms of quality. Before finalizing thedecision, one has to ensure to ask the provider that whether it is possible to test their service.Legitimate long distance providers, in most cases, possess a test procedure in place for the purpose ofquality assurance. Listen for excessive echoing and delayed response. We should feel that we would beable to continue a long conversation with this voice quality without getting frustrated or having torepeat it frequently. 2.1.3. ReliabilityThe reliability of long distance providers’ systems is very important. If the long-distance provider goesdown, our customers’ calls will not go through. Before making the final decision, one can ask them whattype of equipment they have and what kind of fail-over plans they incorporate. Also, we can ask themabout their Post-Dial-Delay (PDD) to different destinations. PDD is, in fact, the time (in seconds) that ittakes for the destination party’s phone to ring after a customer has dialed that destination. 2.1.4. Capacity of ServiceWe may think that our long distance provider can accommodate unlimited voice traffic. Before, signing adeal, one should ask the long distance provider about the capacity of traffic they can handle to aparticular destination. For instance, if we estimate that we will be sending 40,000 minutes of traffic per 6day to India, we should ensure that our long distance provider will not reject some calls because of lack Pageof capacity.
  8. 8. 2.1.5. Finding VoIP Termination/Long Distance ProvidersVoIP Termination Providers sell us traffic to various destinations. When we offer a calling card to call adestination (e.g. to India), we can easily calculate the cost knowing the exact rate of call to India.Assuming a cost of $ 0.0246 per minute to call to India, a calling card that contains 30 minutes to Indiaand is sold for $5, yields a profit of $4.26!It is usually better to select multiple carriers while choosing long distance providers for our calling cardsystem. Most calling card companies provide service to almost anywhere in the world, but it is better tofocus on a particular region of the world. Thus, one can create an A-Z list of long distance providers thatcan send your calls to anywhere in the world. Also, a regional long-distance provider must be chosenthat focuses only on the region of the world that you are targeting. Although A-Z providers areconvenient, their rates are usually higher than regional providers.Major decision factors in evaluating a long distance company are price, quality, reliability and capacity 2.1.6. Finding Access number/DID ProvidersWe need to be looking around for some companies that could provide us with 1-800 numbers or localDIDs in all the states or countries where we would like our customers to have phone numbers, wherepeople could call using any PSTN phone line. The best solution for this is on Voxbone - an exchangeplatform for different VoIP providers to sell DIDs with a low cost per month. 2.2. VOIP ProtocolsVoIP uses RTP for transport, Real-Time Transport Protocol (RTCP) for Quality of Service (QoS) and H.323,SIP, MGCP (Media Gateway Control Protocol/Megaco) for signaling. These VoIP protocols operate in theapplication layer that is, on top of the IP protocol. These protocols that we are using for setting up thecalling card are SIP, H.323, IAX2, PRI/BRI. 2.2.1. H.323“H.323 is a multimedia standard which accommodate a basis for carry voice, video and datacommunications in an IP-based network. The H.323, standards like H.324 (standard for multimediatransport over switched circuit networks) and H.320 (standard for ISDNs) among others. This standard isdefined by an lTU researcher and approved in 1996. H.323 runs on top of TCP in layer 4, and uses TCPfor call setup. Traffic is transmitted on Real Time Protocol (RTP) which runs on top of User DatagramProtocol (UDP).” [17]H.323 defines some clear integral; such as terminals, gateways, gatekeepers and multipoint control units 7 Page(MCU).
  9. 9. Terminals are used for real-time bi-directional multimedia communications, and can be an IP-phone or apersonal computer IP-phone. All H.323 terminals should support H.245 (control channel), Q.931(required for call signaling and setting up the call), Registration Admission Status (RAS is used forinteracting with the gatekeeper) and to support the Real Time Transport Protocol (RTP).Besides, The H.323 terminals can be used to support video or data communications to holding up audiocommunications. Its role for this service (audio communication) in IP-telephony is vital.As we mentioned, the gateway is the interface between the PSTN and the Internet. It can arrangetranslation of protocols for call setup and release, conversion of media formats between differentnetworks, and the transfer of information between H.323 and non-H.323 networks. Also a gateway isable to supply several simultaneous calls between H.323 terminals on the IP network and other ITUterminals on a switched-based network Gatekeepers provide call-control services for H.323 endpoints,such as address translation, admission control, bandwidth management, zone-management, and call-routing services.To allow all the end-users for registering on the VOIP network, they will supply authentication services.The Gatekeepers are the main place for calls in the H.323 network, even though they are an option inthis network.An MCU can be used for multi-conferencing between many H.323 terminals. It controls conferenceresources, and arrangement between terminals for intention of determining the audio or videocoder/decoder to use, and also handle the media stream.A class diagram for VoIP components is shown in figure 2.1. The layer 2 QoS enabled switch supportconnectivity and availability between H.323 components.As we can see in the figure, IP-PBX server accomplishment such as a call processing and handling call setup and routing calls. 2.2.1.1. H.323 Security“The security mechanism in H.323 protects the audio stream as well as the Call Setup (A.931) and CallControl (H.245). H.235 provides security features such as authentication, integrity, confidentiality andsome non-repudiation support in H.323 communications.”*11+ The architecture of H.323 is shown infigure 2.1: 8 Page
  10. 10. Figure 2.1: Class for a H.323 architecture [18]The H.323 applies RTP/RTCP (real-time transport protocol) as its transport protocol which excurse overUDP where encryption is accomplished within the RTP packet by third party hardware or at the networklayer (IPSEC).H.323 can use either symmetric encryption-based authentication or subscription-based authentication.Subscription-based authentication is when a communication occur sharing of a secret key or certificateis mandatory. Certificate-based (symmetric), is password-based (with or without hashing). 2.2.2. The Session Initiation Protocol (SIP)Session Initiation Protocol (SIP) is the IETFs standard for multimedia conferencing over IP. The sessioninitiation Protocol (SIP) is an application-layer control (signaling) protocol used for developing, changingand deciding sessions with one or more assistants.[9]These sessions contain Internet multimedia conferences, Internet telephone calls and multimediasharing.SIP is a text-based protocol, and it’s similar to HTTP and SMTP, to initiate collective communicationsessions between users such as voice, video, chat, interactive games.Signaling allows call information to be transmitted among network boundaries. Session managementsupports the ability to control the attributes of an end-to-end call.SIP is transported over the connection-less UDP protocol. Because of the decreasing state managementoverheads, UDP is preferred over TCP. 9 Page
  11. 11. 2.2.2.1. SIP ArchitectureThe primary of SIP systems are user agents and servers. User Agents (UAs) are a combination of UserAgent Clients (UAC) and User Agent Servers (UAS). The UAC is responsible for initiating a call by sendinga URL addressed invite to the intended recipient. The UAS receives requests and sends back responses.Class of servers:Location servers to get information about a called party’s possible location by a Redirect server or Proxyserver, it will use.Proxy servers are responsible for routing and delivering messages.Redirect servers to inform proxy servers of the user location, it keeps a user location in database.Registrar servers are used to save information about where a party can be found.Figure 2.2 shows the network components and sample message flows for a SIP based network to make acall from a regular telephone number to an IP phone by connecting a Proxy server with a VOIP gateway,and to another Proxy servers. The proxy server is. The proxy server performs on beside of the end usersto ease the call processing. When a call has been set up via the proxy server, the RTP media streamsflow between the end stations.When a user starts a call, a SIP request will send to a SIP server (a proxy or a redirect server). Therequest contains the address of the caller (in the header Field) and the address of the determined callee(in the two header field). SIP architecture is shown in Figure 2.2 on the next page. 2.2.2.2. SIP SecurityThe SIP protocol cannot assign any transport layer security mechanisms by itself, but other protocolssuch as IP Security (IPSec) or Transport Layer Security (TLS) are to provide the needed security for thecomplete message. 10 Page
  12. 12. Figure 2.2: SIP Architecture [18]SIP requests and responses include acute information about the satisfied and designs of communicationof various characters. SIP can support the following methods of encryption to protect confidentiality;End-to-end encryption: Normally, the message is sent encrypted using Public-Key Crypto systems. SIPrequest or response is end-to-end encrypted by breaking up the message to be sent into a part to beencrypted and a short header that will be clear stay.Hop-by-hop encryption: Because header fields need to be visible to proxies (to and via), so, not all of theSIP request and response can be encrypted end to end. 2.2.2.3. SIP ServicesThe services that SIP provides include:• User Location: determine end system to use for communication.• Call Setup: defining and setting up call parameters at both side, called and calling party.• User Availability: determine the readiness of the called party to use in communications.• User Capabilities: determine of the media and media parameters to use.• Call handling: the transfer and termination of calls.Voice gateways usually consist of two parts: the signaling gateway and the media gateway. The signalinggateway uses MGCP (Media Gateway Access Protocol) and Megaco to communicate with the media 11gateway. Both protocols can interoperate with SIP and H.323. SIP transports real time data by usingRTP/RTCP (Real-time Transport Protocol). Page
  13. 13. 2.2.3. Media Gateway Control ProtocolMGCP is a protocol which is produced by the Media Control Working Group and is used for controllingVoIP gateways from external call control essential feature. MGCP systems are building of MediaGateways, Signaling Gateways and Media Gateway Controllers (MGC). MGCP completed the interfacebetween a Media Gateway and a Media Gateway Controller. A place where the gateways areanticipated to accomplish commands sent by Call Agents is a master /slave interface.In this point, control protocol gives the central coordinator authorization to monitor happening in IPphones and inform them to send media a particular addresses.MGCP has presented the idea of connections and endpoints, for setting up voice paths between twoparties. The only thing about MGCP is its possibility to capacity with H.323, SIP, and lagancy telephones.It should be possible for MGCP gateways to do this job with H.323, SIP, and legacy telephones.MGCP Call Control has been secured with using IPsec (with ESP header). Alternatively, a temporaryAuthentication Header (AH) solution should be used. The AH header admit for data sourceauthentication and connectionless reliability of messages passed between the Media Gateway (MG) andthe MGC (Controller), but it does not supply care against replay advance. MGCP suggest using of IPsecfor encryption and authentication. 2.2.4. Inter-Asterisk ExchangeIAX is the Inter-Asterisk exchange protocol native to Asterisk PBX. it supported by number of othersoftswitches and PBXs. It can enable Voip connection between servers beside client–servercommunication.The second version of the IAX is IAX2 which is most commonly.IAX2 is one of the most important VoIP protocols that carries both signaling and media on the sameport. The commands and parameters are transmitted in binary composition and if any extension shouldhave a new numeric code allocated. To communication between endpoints, multiplexing signaling and media flow, IAX2 uses a single UDPdata stream on port 4569. IAX2 covers firewalls and network address translators. This is in contrast toSIP, H.323 and MGCP that use an out-of-band RTP stream to deliver information.“AX2 supports trucking, multiplexing channels over a single link. When trucking, data from multiple callsare merged into a single stream of packets between two endpoints, reducing the IP overhead withoutcreating additional latency. This is advantageous in VoIP transmissions, in which IP headers use a largepercentage of bandwidth.”The IAX and IAX2 protocols were setting up interior session that can use whatever codec that they wantfor transmission. Actually the Inter-Asterisk Exchange protocol supports control and transportation ofstreaming media over IP networks.IAX is used for any kind of streaming media that contain video because it is flexible and also designed for 12control of voice over IP. Page
  14. 14. One of the advantages of IAX is minimize bandwidth for using media transmission. “with specificattention drawn to control and individual voice calls, and to provide native support for NAT (NetworkAddress Translation) transparency.” 2.2.5. Primary Rate Interface and Basic Rate Interface“The Integrated Services Digital Network (ISDN) prescribes two levels of service, the Basic Rate Interface(BRI), intended for the homes and small enterprises, and the Primary Rate Interface (PRI), for largerapplications. Both rates include a number of B-channels and a D-channel. Each B-channel carries data,voice, and other services. The D-channel carries control and signaling information. The Basic RateInterface consists of two 64-kbit/s B-channels and one 16-kbit/s D-channel.The Primary Rate Interface (PRI) consists of 23 B-channels and one 64-kbit/s D-channel using a T1 line,often referred to as "23B + D", or 30 B-channels and one D-channel using an E1 line (Europe/rest ofworld). A T1 Primary Rate Interface user would have access to a 1.472-Mbit/s data service. An E1Primary Rate Interface user would have access to a 1.920 Mbit/s data service.Larger connections are possible using PRI pairing. A dual PRI could have 24+23= 47 B-channels and 1 D-channel (often called "47B + D"), but more commonly has 46 B-channels and 2 D-channels thusproviding a backup signaling channel. The concept applies to E1s as well and both can include more than2 PRIs. Normally, no more than 2 D-channels are provisioned as additional PRIs are added to the group.” 13 Page
  15. 15. 3. Chapter Three: System TopologyBecause of high automation and since internet is the main component and other functions can be doneby one or two devices and servers, typical calling card whole sale business can be handled by 2 or 3people. Here we are going to demonstrate one wholesale company using Sansay SoftSwitch andBillbery - VOIPBilling Server. 3.1. Hardware and DevicesKeyhan Telecom Company consists of four main devices: Switch 3750G, Router 3845, Billing Server anda Softswich. You can see system topology as it has shown below in figure 3.1: 14 Figure 3.1: Keyhan Telecom company Topology Page
  16. 16. On next chapter of the report we will discuss more specifically about Server and SoftswitchConfigurations. Also, you can find Switch and Router configuration in the Appendix. 3.2. Software and ApplicationSoftswitch or software switch is an intelligent platform which is used in voice over IP infrastructure forcall routing, transcoding, signaling, billing and management functions …etc., in IP networks such asphone card companies for cost-effective long distance phone calls.Basically switch is layer two devices with limited capability but here running software used to improveswitch functionality and efficiency.And these days they are faster, cheaper and better replacement devices for traditional hardware basedequipment in telecommunication.Depends on load of work in field of wholesale or retail ,maybe some features of softswitch is handles byseparate specific server and platform like billing servers,These are typical softwitch features and characteristics:Not to mention that every vendor has its own unique attributes.Transcoding protocols:SIP, H323, RTP/RTCP, RAS signaling, and T.38/T.120Media Transcoding:G711, G723, G739Call routing based on:- Destination Gateway Priority- Call arrival time- Minute cost- Operator’s tariff priority- Gateway ability 15 Page
  17. 17. Call Authorization:  Available BW  Available gateway  IP address  Available money in the accountBilling featuresMonitoring online:  Alarm system  Active calls infoCall number transformation:Base on incoming, internal and outgoing prefix.Generally these tasks are handled by softswitch which are divided into two main classes:Class 4 and class 5The main difference between this two is class 4 deals with carriers and class 5 deals with end users. 3.3. Prepaid and Postpaid Calling Cards ServicesPrepaid and postpaid calling card services are different in their billing systems. A prepaid billing modelworks very simple. When a calling card was sold, the service will bill at the time and services aredelivered when the subscriber accesses the retailer’s network. The Cisco prepaid calling card solution isdesigned to give Internet telephony service providers a competitive advantage in the prepaid callingmarket. By tapping the intelligence embedded in IP network components, it allows service providers tocentralize the service application in a single location at a low cost while bandwidth-intensive callconnections are handled at the network edge in Cisco gatekeepers and gateways. The benefit: lowercosts than traditional debit card applications, which are based on service points in large POPs in circuit-switched networks. But in a postpaid model, the subscriber is billed after services are delivered. In otherwords: a seller sells a card with an access number and a PIN number to a subscriber, who can thenaccess the long distance service from any telephone. The long distance service can be delivered via aseller’s own packet voice network, or the seller can partner with a packet-based wholesale terminatingtransporter to deliver the service. 16Prepaid and postpaid calling card services stand for one of the fastest growing types of enhanced voice Pageservices. A selection of consumer segments such as students, business and leisure travelers, expatriates
  18. 18. have propelled the growth of those services. They are especially popular between mobile phone usersas an alternative to the disgracefully high mobile operators’ international rates. For carriers who want torealize more profit from a global long distance network, prepaid and postpaid calling card servicescharacterize a chance to improve margins, direct minutes to the network, and raise customer retention.For service providers that are currently offering prepaid and postpaid calling card services over aswitched circuit network, Cisco packet telephony networks provide a more cost-effective alternative fornetwork expansions or upgrades.Packet voice technology offers a convincing option to the traditional time-division Multiplexing (TDM)switched circuit network and it decrease the cost and time-to-market requirements connected withexpanding voice services such as national and international transport, voicemail/unifiedcommunications, text-to-speech, speech detection, and calling card services. TDM-based services use aleased line and require a long-term financial binder to that exact link, and it (TDM Switch) also presentsmajor first cash outlay, requiring a lengthy time period to get investment payback. The need to getfaster investment payback has led some providers to add fees for calling card activation or connection tomake up the difference.Cisco offers a high quality and practical solution for prepaid and postpaid calling card services that isdeployed via packet voice technology. The Cisco Voice Infrastructure and Applications (VIA) solutionincludes key features and attributes such as:  A telephony user interface similar to familiar Public Switched Telephone Network (PSTN)  Card services applications  Cost-efficiency in equipment and bandwidth  Card recharging  Balance transfer  Personal identification number (PIN) change.  Support for multiple languages.  Support for multiple-company branding or announcement messages on the same network.The basic structure of Cisco VIA are the Cisco media gateway, call control (Cisco gatekeeper, Cisco SIPProxy Server, Cisco PGW 2200 softswitch, or partner call control), IP-to IP-interconnect (using the CiscoDirectory Gatekeeper or SIP Proxy Server) and operations support systems (OSSs) to manage andprovision the entire network. These allow Cisco VIA to support the following services such as Prepaidand postpaid calling card services, national and international transport, termination services forapplication service providers (ASPs), voice mail and unified and communications and dial access.Cisco VIA has been deployed in more than 80 countries and by hundreds of service providers whichprovide flexibility to use the Media Gateway Control ProProxy Server, or provide the ability to use the Media Gateway Control Protocol (MGCP) to interface to a 17Cisco softswitch. Page
  19. 19. The Cisco VIA solution offers so many profits, such as lower transportation and operating costs comparewith other industry offerings, and It offers industry-leading voice quality, fixed reliability, and scalabilityto suit a variety of network sizes, and protocol flexibility, and it enables service providers of any size andlocation anywhere in the world to contend in the calling card services sell. 3.3.1. Prepaid and postpaid Service explanationA prepaid or postpaid calling card service can be presented in sell or wholesale models. The wholesalecarrier manages the calling card service on its international infrastructure, that the most prepaid callingcard services get benefit from wholesale model. In other hand the retail service provider offering to usermarkets and brands the calling card service. For both prepaid a postpaid calling card services also offersubscribers continuing permit to the long distance network. Similar to prepaid calling cards, the postpaidcalling card service is often hosted by a wholesale bearer to increase advantage. The most differencebetween prepaid and postpaid calling card services is that service authorizations in the postpaid modelare not attach to call rating and services do not expire, except in the case of a limited-credit postpaidservice because of the call rating does not happen in real time.Prepaid and postpaid calling card services provide carriers with an opportunity to improve margins thatthe price per minute billed is bigger than residential or dial services and also increased minutes isminutes to the packet telephony network, and increase customer retention that the prepaid cardservices delivered on a VOIP network with offerings speed dial and voice email that has much lower costin PSTN services. 3.4. Solutions 3.4.1. Advantage of Cisco VIA Solution in Calling Cards ServicesOne of the most advantages of Cisco VIA solution to service provider is in the calling card servicesmarket. It also offers the greatest choice for interconnection through its Cisco Service CarrierCommunity program. This program helps service provider develop traffic on their networks.As we know this business is very competitive and price sensitive, with agile customer loyalty. Indefinitelyto achieve the largest volume, service providers should offer lowest prices services. The key to achievefinancial success in industry is to adjust a low price service with a low cost infrastructure and also lowcost operation, and differentiated aspects. These articles can be able to reach with separate equipment,billing systems and accommodation call paths, that Cisco VIA solution took this advantage. The IPtelephony technology in the Cisco VIA solution perfectly addresses the needs of a service providerbecause it is a low cost infrastructure solution. The basic of this service enabling solution include Cisco 18AS5000 universal gateways using Cisco IOS Software, the Cisco PGW 2200 PSTN Gateway, and Cisco2600 and 3600 series routers (refer to Figure 2). The Cisco solution offers service providers one more Pageimportant advantage. The IVR system, a critical part of any calling card application and normally a
  20. 20. significant cost factor in delivering the service, is included in Cisco AS5000 universal gateways. Thisintegrated feature substantially reduces the costs of providing the service. An overview of a network,working with Cisco Solution and Gateways are shown in figure 3.2: Figure 3.2: A network contains Cisco Solutions and Gateways [19]“In conjunction with these products, Cisco Service Provider Solutions Ecosystem partners provideaccounting and billing applications that complete the Cisco prepaid and postpaid calling card servicesolution. These partners include dig quant Systems, Mind CTI, Portal Software, and Primal. Theapplications these partners provide enable a rich set of options that enhance revenue, createOpportunities for service distinction, and mitigate cash flow risk. Setup, recurring, and usage-basedcharges may be customized to accommodate a variety of regional, cultural, socioeconomic, servicequality, and market trend shifts. “Session Initial Protocol (SIP) and H.323 are protocols by Cisco solution VIA. 3.4.2. Cisco SP Voice SolutionThe Cisco SP Business Voice Solution offers deliver managed voice service on IP Communication solutionto service provider, and also products to any size of business. With this kind of solution, service providercan arrange a scalable, reliable, and Voice over IP infrastructure can able them to offer a selection ofmanaged voice services which can work with several end-customer adding to options that include CiscoCallManager, Cisco CallManager Express, legacy private branch exchanges (PBXs) with VoIP gateways,integrated access devices (IADs), and remote Cisco IP phones.What services are enabled by the Cisco SP Business Voice Solution enable services to its customers inany small and enterprise size such as (Figure 3.3): 19 Page
  21. 21. Figure 3.3: Business Voice Services Description [19]The flexibility of the Cisco SP Business Voice Solution architecture will also enable service providers togain on arriving profits opportunities from managing improved IP applications. These include IP phoneExtensible Markup Language (XML) applications and IP customer contact services, IP conferencing, andothers. 3.4.3. Global Long Distance SolutionGlobal Long Distance Solution provides the ability to offer wholesale and international transit services.Based on IOS technology, the solution provides full range of TDM interfaces, such as R2, PRI and SS7.The solution offers enhance services such as:  Prepaid and Post Paid Calling Card Services  Unified Messaging  Dial Access Services  Seamless AVVID Integration  Voice VPN ServicesAll of these services can be leverage from the same common voice gateway, running Cisco IOStechnology. As for the value proposition, it is the only solution on market that brings such services in acost effective platform. The solution scales from a few E1’s to tens of thousands E1s. The robust carrier 20class gateways provide support for both H.323 and SIP on a call by call basis using the same load. This Page
  22. 22. means with Cisco technology, service providers can deploy a robust H.323 network and still support SIPtraffic terminating from ASP providers such as Microsoft XP (Shown in Figure3.4). Figure 3.4: Partnership of Cisco and Microsoft [2] 3.4.4. Integrate Transport SolutionWith the integrated transport solution (ITS) we deploy 8850 ATM switches with the Voice InterworkingServices Module (VISM) – without a softswitch. The switches are connected to either an ATM or IP core.The ITS is typically used by PTTs or Mobile operators to reduce backhaul cost in configurations wherethey don’t need or are not ready for Softswitch. The solution give carriers an immediate 4X cost savingon backhaul costs (using compression) and requires no changes to the carriers existing circuit switches(its completely transparent to them). The added bonus is that carriers can easily add a softswitch to thisarchitecture to take advantage of a “switched” transit solution. The 8850/VISM is softswitch ready (canbe controlled with MGCP). 21 Page
  23. 23. 3.4.5. Wireless Transit SolutionThe wireless transit solution is a slight variant on the wireline transit solution. We position the wirelesstransit solution with mobile operators who are looking to offload their inter-MSC circuits from TDM ontoVOIP or VoATM. Given that many mobile operators in AsiaPac are running near or at capacity on theirnetwork, we also see the incorporation of Gateway Mobile switching center functionality into thesoftswitch as another major cost saving benefit for the carrier – through offload of their legacy circuit-switch Serving Mobile Switching Centers. (Gateway MSC functionality - softswitch queries the HomeLocation Register on incoming calls and delivers the call to where the mobile subscriber is currentlyroaming). In the future, we see the Gateway MSC softswitch evolving to support full “Serving” MSCcapabilities (I.e. actually controlling the radio equipment. Wireless infrastructure is shown in Figure 3.5: Figure 3.5: Wireless Transit Solution infrastructure [2] 22 Page
  24. 24. 4. Chapter Four: Billing Server and Softswich 4.1. Billing Server: Jerasoft VoIP Carrier Suite (VCS)In whole sale solution when company deals with large number of calls billing task is done by anotherserver called billing server as you can guess from the name it’s pretty straight forward function: countthe minutes and provide the invoice base on the rates. Jerasoft GUI overview is shown in Figure 4.1.Like softswitches every vendor has its own features but the main functions are the same:  User management: Administrator, reseller, group seller, administrator…  Prepaid service: - Real Time Balance Deduct - Subscriber/Reseller Recharge - Effective Date/Expired Date - PIN Code Generate and Consume - Recharge Log  Postpaid Service - CDR Report - Call Detail Record Storage - Effective Date/Expired Date  Flexible Rate Plan Support - Up-to 5 Charge Segments per Rate Prefix - Effective Date/Expired Date - Programmable charge unit, amount and cycle - Call Screening - Support Per Call , Holiday & Night Time Charge - Longest Prefix Match - Free Monthly and Deductible Minutes based on Prefix - Database support 23 Page
  25. 25. Figure 4.1: Jerasoft system GUI overview Like other vendors BillBery solution supports postpaid and prepaid billing services, including calling cards, Caller-ID, dynamic call routing base on rate management which means chose the best rate at the time. It is used for whole sale, retail and even call shop with capacity of large number calculation each month. And also it’s compatible with other vendors such a Cisco, Sansay, Asterisk Nextone… Bilberry VCS package has different modules: Billing, Routing, retail and Rate control. 4.2. Softswitch: Sansay, the VSX-Integrated Multimedia SubsystemThe VSXi is designed to enhance Sansay’s leadership position in the Access and peering SBC marketplace.By integrating the best of the VSX and SPX products and including important enhancements, the VSXiexpands the addressable applications and provides higher network availability. Although the applicationdiagrams look very similar to the existing product set, the VSXi uses a very different internal architecturewith a new Data Base format and new SIP stack. A new additional hardware platform might be required 24to minimize the downtime associated with product migration. Applications include: Page
  26. 26. • Retail subscriber access to feature servers• Retail SIP trunking applications• Wholesale peering applications• Combinations of the aboveMonitoring ApplicationThe Sansay Network Session Monitor is used to obtain near real time call performance statistics andprovide historical performance reporting. The NSM is capable of monitoring all network resources andcharting the call activity in real time. This can be used from a NOC workstation or a personal computer,to keep track of the health of the network or for post analysis for vendor and route performance. This application is provided on the system documentation CDROM which is packed in the shippingcontainer with your server.VSXiThe VSXi builds upon the Sansay product family and will replace the existing products with three newcomponents. The VSXi is the integrated system, which will operate in a stand-alone or a HighAvailability pair (HA). In this configuration, it will address the small to medium single locationapplication with full media switching support.VSXrThe VSXr is a subsystem of the VSXi and controls the call routing for the sessions. The VSXr uses a DataBase to analyze the session parameters and Least Cost Routing tables in order to find the properoutbound call path. The VSXr is used for larger applications and is roughly analogous to a DRX.VSXcThe VSXc is the call processing subsystem for the VSXi and controls the interface to all devices withregard to signaling and media flows. VSXc will write and store all Call Detail Records (CDRs) to beretrieved by the billing application. The VSXc is used for large or distributed applications and providesthe Virtual IPs for all services. 4.2.1. VSXi Primary Features  Call Routing – The VSXi improves the previous VSX routing allowing routing with respect to 25 FQDNs or IP addresses. This provides easy service partitioning on inbound calls. Calls can arrive Page
  27. 27. on one Virtual IP address and depart on another. This allows consolidation of network interfaces to be accomplished independent of the routing within the system.  Protocol internetworking – Supports any H.323 fast start to SIP call Inter-networking. Also, provides Gatekeeper inter-networking to SIP.  Service Ports – Introduces the concept of Service Ports. Service Ports can be thought of as connection points to carriers, SIP peers or end access devices like phones or small gateways. Service ports are assigned a specific, unique virtual IP address and UDP port combination which will be used by the VSXi to provide the services for customers and vendors.  Service Ports are used for signaling and media transmission into and out of the VSXi.  Video Support- VSXi transparently supports video for video phones. This requires direct media to be set to yes.  High Availability – A fault tolerant, non-stop cluster. System processes, poll each other constantly. During a failure, the backup system assumes the virtual IP address of the failed system. Connected calls will not be dropped during failover. When calls are cleared they are logged in redundant CDR files. MTBF for servers are 36,000 hours.  Local Number Portability- The VSXi provides easy access to LNP services through intelligent handling of SIP redirect messages. Following a DIP of a ported number, the VSXi will route the call based on the LRN (location routing number) number returned and write that number in the CDR for that call.  Cluster License Zones – The VSXi has the ability to use several systems for redundancy, but still provide accurate Call Admissions Control on Trunk IDs and Gateways. This provides scalability and geographic redundancy while maintaining complete control over traffic patterns. The CLZ is configurable in the GUI.  Network License Zones – The VSXi is able to provide license sharing across many different areas of the network, even when the systems are operating entirely independently in terms of configuration. This permits the optimization of licenses regardless of traffic patterns or configuration differences in systems around the world. The NLZ system IP address must be licensed in order to use this feature.VSXi uses an enhanced software architecture, improved Data Base and new SIP stack, to combine thebest features of the SPX and VSX in one unit. It can operate in a single, paired or clustered environment.Many features requested for in the VSX and SPX have been designed into the VSXi.• It is a carrier class Session Border Controller, controlling routing in VoIP networks• Provides Denial of Service (DOS) and DDOS protection 26• Centralizes routing tables Page
  28. 28. • Centralizes CDRs• Intelligent route hunting• Real time performance monitoring• Protects all internal IP addresses• Reduces load on soft switches and gateways• Originates media and signaling• Advertises only one IP address using topology hiding• Supports H.323 Gatekeeper to H.323 Gateway• H.245 tunneling conversion• Full SIP method support• Any H.323 to SIP conversion• Can perform Registrar duties to reduce load on feature servers. Subscriber to subscriber calls can berouted directly via the VSXi• Provides video transparency for video phone support, when media is direct 4.3. Subscriber Access ApplicationRetail subscriber access is for Vonage-style applications and hosted PBX networks with high numbers ofsubscribers. Key features for this application are:• 250,000 active subscribers and up to 8000 sessions per pair• 16 pairs per cluster resulting in a maximum capacity or 4 million subscribers in a cluster with 108,000sessions.• DOS and DDOS protection• FQDN, name based stateful forwarding• Topology hiding 27• NAT traversal Page• Easy GUI Management
  29. 29. • License Sharing Zones providing real time sharing of network resources, lowering capital expenses andoperating expenses.The VSXi will provide routing to the TIDs if necessary, bringing the Access Application into completealignment with the Peering (LCR) application. Network routing can also be accomplished by the featureservers, which addresses tier one applications with a more generic SBC model. The subscriber accessapplication can also be supplemented by the SIP trunking application as the carrier grows in one marketor the other. VSX infrastructure is shown in Figure 4.2: Figure 4.2: VSX infrastructure 4.3.1. SIP Trunking ApplicationWhen the VSXi is used in a SIP trunking application, it may also be used as a hosted IP PBX allowing onedeployed platform to perform several functions. Key Features are:• 100,000 trunk groups and 8000 simultaneous sessions per VSXi pair, with a maximum of 4 millionsubscribers and up to 108,000 sessions per 16 pair cluster• LNP, CNAM, RADIUS, and Teleblock access 28• TCP and TLS support for connections to Microsoft and other IP PBXs Page
  30. 30. • Ability to have IP PBXs register directly to the VSXi for small business applications, with changeable IPaddresses• Can route E911 calls to service providers with the advanced PID-IFLO headers and act on the 302redirect to the appropriate Public Safety Answer Point (PSAP) number.• A future release of the VSXi will incorporate Datagram Transport Layer Security (DTLS) as an accessprotocol which will become more useful as IP Multimedia System (IMS) user endpoints becomeavailable.It is not necessary to provide feature servers in a pure SIP trunking application, but the use of featureservers for true retail Class-5 services is a simple configuration change in the VSXi with the use of FQDNforwarding in the route tables. VSX Functionality is shown in Figure 4.3: Figure 4.3: VSX Functionality 4.3.2. VSXi Wholesale ApplicationIn a wholesale application, the VSXi provides the ability to have high speed sophisticated routing in ascalable, manageable, and highly reliable network. The VSXi continues the advanced routing of the VSXwith intelligent DNIS/ANI relational routing and alternate route choices. There are a total of 32alternate routes using the 8 routes and the route link table. Key features are:• 100,000 trunk groups and 1000 different route tables 29• 1000 calls per second and supports 8000 calls with full topology hiding Page• Internetworking with SIP and H.323 available on all calls
  31. 31. • SIP, TCP/TLS calls are internetworked to normal SIP or H.323 routesThe VSXi allows carrier interconnects which require different IP addresses for different services. It hasthe ability to send calls to a particular vendor IP address from different TIDs, using different sending VIPaddresses. Wholesale VSX functionality is shown in Figure 4.4: Figure 4.4: Wholesale VSX functionalityNote: VSXi does not support H.323 Slow Start ProtocolNote: VSXi will not provide H.245 to RFC2833 conversion, but does use the H.245 to SIP INFO method 4.4. VSXi hardware• Dual Gigabit and 10/100 baseT Ethernet interfaces with an option for up to 4 extra 10/100/1000interfaces• 1U servero 1000 active calls (H.323 or SIP with media) 30o Non redundant Pageo AC only
  32. 32. • 2U servero 8000 active calls (H.323 or SIP with media) per chassiso Redundancy on fans, power supplies and RAID driveso AC or DC power• 2U NEBS for Central Office Applicationso Same performance, but NEBS 3 certifiedo Redundant AC or DC power supply optionServer rear view is shown in Figure 4.5: Figure 4.5: Sansay and Billbery Servers 31 Page
  33. 33. 4.5. VSXi ConfigurationThe VSXi features a Graphic User Interface. All configurations can be accomplished using this GUI.Open a browser and enter the Sansay default private side, IP address or a preconfigured IP address. Thedefault is https://10.10.10.100:8888/ this is the private side default IP address (Figure 4.6). Figure 4.6: Login InterfaceOnce you have logged in, you will see the GUI main screen. You will want to edit your network settingsnow to facilitate installation into your network. (Figure 4.7 shows Sansay overview. Also, all the systemtabs are shown in the following Figures of this chapter.) 32 Page
  34. 34. Figure 4.7: Sansay GUI overview4.5.1. Sansay GUI Major Tab: System (Figure 4.8) 33 Figure 4.8: System Tab Page
  35. 35. o Basic Tab (Editing Network Settings)Click the system tab to begin to edit the Network Settings. Click the Edit Network Connections tab to setthe initial network configuration for the VSXi. (Figure 4.9) Figure 4.9: Basic TabLAN Interface 1 and 2Specify the IP address of the VSXi system for both the public and private LAN. This will enable you tomanage the VSXi from any remote system. LAN 1 corresponds to Ethernet 0 on the System Stats page.Set the subnet mask for both private and public LAN. If you are not sure of these addresses, check withyour network administrator.Set the IP address of the default gateway for the public and private LAN. If you are not sure what toenter, contact your network administrator.Set the Network Mask. This is used to separate the public and private LANs. Commonly the Netmaskvalue is 8, 16, or 24 and is dependent on the number of networks in used on the private network side.This value indicates how large the private address network is, based on the length of the network bitswithin the address. If you are not sure of these settings, ask your network administrator. This valuedetermines when the VSXi will transmit a packet onto the private LAN. It can be considered a staticrouting option by which packets are checked before being transmitted onto the public interface. If thedestination IP address is not within the Private Address contiguous space, the packet will be sent on thepublic interface. Special static routes can be added for VPN tunnels or other customizations. 34 Page
  36. 36. Edit System PageClick this tab to edit system name, CDR information, payload port, SIP port, H.323 information, NTPinformation, DNS information and system time, date and time zone configuration.AliasSpecify the name for the VSXi system. The alias can be up to 40 characters in length and can includecommas, semicolons, spaces, periods, hyphens, underscores, the @ symbol and questions marks. Youmay choose any name you want for alias within the guidelines above.CDR System NameThis name can again be whatever name you wish or can be left default as Sansay. The CDRs can be usedto bill customers and troubleshoot problems with your routing or carrier suite.CDR File IntervalSpecify the number of seconds between file writes. CDR files are constantly written. CDR files will beautomatically purged every 14 days, if not manually deleted.CDR PasswordAdd a CDR password. This will be used to retrieve CDRs. The user name is fixed as cdr for retrieval.Local Payload Port StartSpecify the starting UDP port number to be used by the VSXi for payload. The default UDP port startingaddress is 10000.DNS Server 1 and 2Set the IP address of your DNS servers. If you are not sure what to add here, ask your networkadministrator. Even though all TIDs may be configured with IP addresses, there may be FDNs in theContact field of the messages. It is highly recommended to have these DNS fields configured.Gatekeeper IDSet the IP or FQDN of the Gatekeeper for H.323. This is the name that the other Gatekeeper will seewhen the VSXi signals a call outbound to them. The other Gatekeeper will likely need this address toaccept calls from the VSXi.NTP Server 1 and 2Set the IP address for up to two network time protocol servers. These are used for reference to external 35time standards. Due to internal server clock drift, it is highly recommended that NTP servers be set.These reference times will affect your CDR information. Page
  37. 37. Current Time, Date, and Time ZoneSet the current time, date and select the time zone for your area. Don’t forget to select the Submitbutton to invoke your changes. (Figure 4.8) Figure 4.10: System TimingNote: This will cause a system reset due to the need to maintain proper time stamps. o SNMP Servers TabIf you would like to send alarm traps to one or more SNMP servers, click the SNMP Servers Tab from theSystem Tab on the main screen. Click Edit to add or edit a tab. Input the FQDN or IP address of theserver/servers the version of SNMP, the string and severity level. If the VSXi is configured with SNMPservers, it will send SNMP traps to the configured servers when there are any system related problems. 36Please see the appendix for SNMP trap definitions. (Figure 4.11) Page
  38. 38. Figure 4.11: SNMP Tab o Advanced TabFrom this tab, you can save your configuration and restore it at a later time. This is very helpful in caseof server failure or if someone makes changes to the configuration that cause critical call failures. Youshould keep a good copy of your configuration saved for just such possibilities. To save or restore yourconfiguration, select the System tab, then Advanced, then choose Save Configuration or RestoreConfiguration. (Figure 4.12) Figure 4.12: Advanced TabSystem OfflineSystem Offline will take the system offline. You will receive a warning: “System will not take any newcalls. Please restart the system to get back to normal operation”. You will then be given an opportunityto cancel or take the system offline. System offline will cause the system to reject any new inbound 37calls, but will not affect any in progress calls. No new outbound calls will be allowed, so as in progresscalls are completed the system can then be restarted. Page
  39. 39. Reboot SystemReboot System will allow you to reboot the system. This will terminate all current calls. You will begiven an opportunity to cancel or Reboot Now.Upgrade CodeTo Upgrade Code, select the Upload Code Upgrade button. You can type the file name or Browse to thefile. You will be given the opportunity to cancel or submit. Once the new file is selected, click theSubmit button to begin the upload of new code. After the upload is complete, you will be asked torestart the system in order to activate the new code. You may restart immediately or choose to restartat a more convenient time, using the Restart Software button. The new software version will be loadedduring the restart.Restart SoftwareThe Restart Software button will restart the application software. This will terminate all current calls.You will be given the option to cancel or Restart Now.Shutdown SystemThe Shutdown System tab will allow you to shut down the server. This shuts down all processes andparks the disks. You will be asked to place the system offline before proceeding to allow all active callsto be terminated gracefully.Note: The System Shutdown command will require someone to be onsite to power the server back on.Clear Log FilesThe Clear Log Files tab will immediately clear all the log files from the server. You will not be given awarning. Once the Clear Log Files Tab is selected, all log files are cleared!  Major Tab: Service PortsService ports can be thought of as points of connections to carriers, SIP peers or end access devices likephones and gateways. They specify a unique LAN interface Virtual IP address and UDP port combinationwhich will be used by the VSXi in order to provide the services for customers and vendors. Service portsare used for media and signaling into and out of the VSXi. There should be no services provisioned on 38the static IP addresses of the system. All trunk ID Resources need to be assigned to a specific ServicePort. Service Ports should be configured prior to TIDs. TIDs are configured on the Resources Add page. Page
  40. 40. Service Ports combined with Routes and Resources, provide intelligent connections between class 5VoIP switches and VoIP termination partners. This provides the call control softswitch with a safe,secure connection to partners over the internet. An equally important feature is the protocolconversion provided by these ports. Each resource using a Service Port can be configured as H.323 orSIP. In general, SIP ports are configured on UDP port 5060 or TCP port 5061. H.323 is usuallyprovisioned on TCP port 1720.There are two types of Service Ports, Access and Peering. Access provides for a connection to a phone,gateway or IP PBX, which needs to register and may have changeable IP addresses. Peering is forcustomer or vendor traffic with fixed FQDNs or IP addresses. This requires calls be routed according to aroute table. Virtual IP addresses are also configured on the Service Ports Tab.Add a Service PortTo add a Service Port, select the Service Ports tab from the Main GUI page. Then select the Add tab. Inthe trunk ID configuration, you will be required to assign the TID to use a specific Service Port. Routeentries will point calls to specific outbound Resources which control the outbound Service Port to beused. Route configuration will be covered in a later chapter. (Figure 4.13 and 4.14) Figure 4.13: Service Ports Overview 39 Page
  41. 41. Figure 4.14: Service Port AddIndexService Port Index is a unique number to identify your Service Port. You may choose any number youwish for this index. Resources will be assigned to use this number. Valid range is 0 to 1000.AliasThe Alias is a name you give to your Service Point. This can be important if you make your Aliasdescriptive, so you do not have to look up IP addresses, when you are setting up a route.Ethernet InterfaceEthernet Interface reflects either eth0 or eth1. Ethernet 0 is the public side and 1 is the private side.Virtual IPVirtual IP Address is the IP associated with the connection to this port. Select this from the drop downlist. Virtual IP addresses are configured under the Service Port Tab, then select Virtual IP.Service TypeService Type is SIP, or H323 and must match the connection to this port.PortThis is the port that the VSXi will listen for inbound traffic. The TID configuration will identify the far end 40device port which will be used. SIP default ports are 5060 for UDP and 5061 for TCP. H.323 gatewaydefault TCP port is 1720 and gatekeeper default TCP port is 1719. Other ports may be used in your Pagespecific network.
  42. 42. Port TypePort Type choices are UDP, TCP, TLS, or DTLS. Again this must match the devices you connect to withthis port.Resource TypeResource Type choices are Access, Peering or Dynamic Peering. Access is a connection to a subscriber orregistering Gateway. A Peering connection will cause the calls to be routed according to the RouteTables assigned to the Resources which use a specific UDP port. Dynamic Peering is for use where IPaddresses may change. For example: behind a firewall or with DHCP. Dynamic Peering requires thetrunking gateways to register to the VSXi and supports SIP only. The gateways need to be configured inthe Subscriber Resource Table. You will need to assign passwords so they may register with the VSXi.Unlike normal Subscribers, the VSXi does not match the user name to the URI in order to send calls. TheSubscriber User name is only used to validate the gateway.Note: Any TID using the deleted Service Port will be set to Service Port 0. (no service port) and will stopprocessing calls from this Resource Port. The user must delete or change the resource which uses thisservice port before deleting the service port itself. An attempt to delete a service port used by anyresource will result in an error message with a list of resources that the user has to edit first. o Virtual IP Tab Figure 4.15: Virtual IPAdd a VIP to your system. Press the Add button. Input the IP address and select the appropriate LANinterface. (Figure 4.15 and 4.16) 41Note: This will activate a new VIP on your system. Please insure that the VIP is not in use already. If theVIP is being used already, service on other VIPs may be impacted. Page
  43. 43. Figure 4.16: Virtual IP add  Major Tab: Application Servers o Radius Servers Tab Radius Servers are principally used for storage of CDR information, but can also be used forauthentication. The VSXi System Page allows you to configure Radius Servers if they are being used. Youcan configure up to 4 Radius servers for redundancy. Each radius message is sent to the 4 servers innumerical order. Each server will be attempted multiple times set by the Radius Max Resend. TheRadius Server Shared secret has to be coordinated with the radius system, as well as the UDP used forAuthorization and Accounting. Usually the defaults for the UDP ports are fine. The Radius Resend timecan be programmed as well. Depending on the load on the server the resend timer of 3 seconds shouldbe sufficient. The VSXi has a special Radius Recovery feature which increases the reliability of the radiusfeature. If a specific Radius message is not acknowledged by any of the programmed radius severs, theVSXi will store the message in an error file and will automatically resend the messages to the radiusserver when it is available again.Add a Radius ServerTo Add a Radius Server, from the main GUI page, select App Servers, then the Radius tab, then Add.(Figure 4.17 and 4.18) 42 Page
  44. 44. Figure 4.17: Radius Servers Figure 4.18: Radius GroupGroup IndexEnter a Group Index number between 1 and 16. This identifies which Radius Server applies to whichResource. 43 Page
  45. 45. Group PolicySelect a Group Policy from the drop down list. Choices are round_robin or top down. This refers to theway the VSXi will access the servers. Top Down means VSXi will always start with the top of the list andwork its way down only if the first server is unavailable. Round Robin means the next server to beaccessed will be the one on the list, just below the last one used.Server Index, FQDN, Server Secret Password, Auth Port, Accounting Port, Resend Period, Max ResendsEnter a Server Index, FQDN, Server Secret Password, Authentication Port, Accounting (CDR) Port, ResendPeriod, Max number of resends and Stop Records Only. If stop records only is enabled, the VSXi willsend only the Radius stop record to the server. The call Start Records will not transmitted if stop isenabled. In most cases the Start records are not needed for call billing. o CNAM Servers TabCNAM is used for outbound call treatment to query a DB server or server that will present the displayname used for advanced Caller-ID. The VSXi will send a SIP Subscribe message with an ANI to theservers, which will respond with the Display Name of the caller. The VSXi will then add this informationto the Proxy Asserted Identify field and From field of the outbound call. If the CNAM is already presenton the inbound call the VSXi will not query the server. The VSXi uses the PAI or RPID lines to ascertaininbound CNAM presence. (Figure 4.19) Figure 4.19: CNAM ServersGroup IndexEnter a Group Index number between 1 and 16. This identifies which CNAM Server applies to whichResource.Group PolicySelect a Group Policy from the drop down list. Choices are round_robin or top down. This refers to theway the VSXi will access the servers. Top Down means VSXi will always start with the top of the list and 44work its way down only if the first server is unavailable. Round Robin means the next server to be Pageaccessed will be the one on the list, just below the last one used. (Figure 4.20)
  46. 46. Figure 4.20: CNAM Group o Local Number Portability Servers (LNP) TabLPN is used to check the dialed number to see if it has been ported. (moved to a new service provider)This is done using an Invite to the LNP server which will return a 302 Redirect message. This messageincludes the Location Record Number (LRN). When the LRN is available from the server, or on aninbound call, the VSXi will use it for the route lookup instead of the DNIS. Both LRN and DNIS areforwarded to the next call leg. o Teleblock Servers TabTeleblock servers are used to check if the dialed numbers are on the U.S. Do Not Call List preventingtelemarketing calls. This is applied to inbound calls on specific TIDs and should only be enabled on TIDswhich connect to call centers.Group IndexEnter a Group Index number between 1 and 16. This identifies which Teleblock Server applies to whichResource.Group PolicySelect a Group Policy from the drop down list. Choices are round_robin or top down. This refers to theway the VSXi will access the servers. Top Down means VSXi will always start with the top of the list andwork its way down only if the first server is unavailable. Round Robin means the next server to beaccessed will be the one on the list, just below the last one used. 45 Page
  47. 47.  Major Tab: ResourcesA Resource is any device that will send or receive calls to/from the VSXi. It could be a small IP gatewaywith only one analog port, a PC soft client, or a gateway as large as Cisco 5850 or Sonus GSX9000. If adevice is not configured in the VSXi as a resource it will not be able to generate or receive calls from theVSXi. A Resource can also be a range of IPs, specified using an IP address and a mask range. It is possibleto allow calls from an entire B-Class network using an IP address such as 69.63.193.187/16. This Trunk IDwould encompass all IPs between 69.63.0.0 - 69.63.254.254. To Add a Resource, select the ResourcesTab from the main GUI page. (Figure 4.21 and 4.22) Figure 4.21: Resources 46 Page
  48. 48. Figure 4.22: Adding ResourcesResource TypeSelect the resource type, from the drop down, Peering, Access, or Dynamic Peering. Peering areconnections to a carrier partner. This could be a gateway or IP PBX. Dynamic Peering is the same aspeering, except usually behind a firewall or other device that requires NAT. This device can register foritself only, not on behalf of subscribers. Access is a single line, like a SIP phone or small gateway. Thedifference is each access device registers on its own for a single line. Resource Type fields enable you toselect the resource protocol type and set operational parameters such as Service State, directionalproperties, network address translation (NAT) and media handling.ProtocolSelect the protocol you will be using. Drop down choices are SIP, H.323 GK, H.323 GW or ENUM. The 47protocol selection is used to control the outbound call request protocol. All calls sent to this resource Page
  49. 49. will use the configured protocol. However, inbound calls are accepted with SIP GW, SIP Proxy, H.323GW, H.323 GK, or ENUM Server protocols.Port AddressAdd the port address you will use. The default is 5060 for SIP. You can leave this default if you wish.Trunk IDSpecify the desired trunk ID for the resource. This will be the default trunk ID. For numbering purposes,the TID range for Resources can be 1-99,999. The TID number must be unique and is used inconfiguring the route tables to direct calls. If multiple TIDs are configured for the same IP address, oninbound calls, they must have a Tech Prefix assigned to differentiate between TIDs. For outbound to theIP address, a Tech Prefix is not necessary since the route entry will determine which TID is to be used.NameUse this field to specify the resources alias name. You can enter a name of up to 40 alphanumericcharacters, including periods and hyphens. This field is optional.Company NameUse this field to enter a label to identify the gateways owner. You can enter a company name of up to40 alphanumeric characters, including periods and hyphens. This field is mandatory.Route TableSelect the appropriate Route Table from the drop down list.Remote PortThis is the SIP port to be used inbound. This is the UDP or TCP port that the remote device uses forreceiving signaling traffic from the VSXi.Service PortUse this field to select a Service Port previously configured. The TID will only use this VIP and port tosend or receive traffic. Media will be sent on the same VIP, but a different port.Aggregate CapacityEnter the capacity in sessions for this resource. This is the maximum capacity in sessions allowed for thisresource, similar to the effect of CPS limit.Aggregate CPS limit 48Use this field to set calls per second allowed on this resource. This works in conjunction with the PageAggregate Capacity to limit Sessions and CPS for a resource. Resources can be controlled on a calls-per-
  50. 50. second basis. CPS limiting measures the CPS (inbound and outbound) on the TID and rejects calls thatexceed that value. Any value from 1 to 1000 can be entered. Inbound calls that exceed the CPS range arerejected, while outbound calls roll over this route selection and are rerouted if an alternate route isconfigured. On routes which are commonly over used during peak periods it is beneficial to limit theCPS to a reasonable number. This will prevent the VSXi from overrunning the terminating resource aswell as wasting resources sending large quantities of calls to devices which cannot accept them.Group PolicyDrop down choices are round robin or top down. Round Robin means each new call will start at thenext resource after the one used in the previous call. Top down means the call will always start at thetop of the hunt list and work down if necessary.Digit Mapping TableSelect either, no translation or one of the mapping tables from the list.Max Call DurationSet the maximum call duration for any one call. Limits are 10 to 86400 seconds.Payload TypeRTP payload type for RFC2833 packets usually 101.RTS/TOS DiffservEnter in hex the diffserv bits for flow control. All signaling and media for this TID will use this value.DirectionEnter whether this resource will be inbound, outbound or both ways from the drop down box.Determine if the gateway will send calls to the VSXi, receive calls from the VSXi, or both. Select In todesignate the gateway as an ingress gateway to the VSXi. Select Out to designate the gateway as anegress gateway from the VSXi. Select Both for bidirectional operation.The VSXi determines inbound calls based on the IP Address and Tech Prefix. If Tech Prefixes are notused, inbound calls will find the first TID number that matches the IP in ascending order. If you haveinbound TIDs with the same IP as outbound TIDs, they should be numbered lower than the outboundTIDs.Service StateEnter in-service, block or trace. Determine the operational state for the gateway. Select In Service to 49bring the gateway online, Block to disable access to/from the gateway, or Trace to enable the tracingfunction. Page
  51. 51. Note: Only trace 1 resource or route at a time. This will ensure that the call being traced will be fromthe desired resource. The VSXi will only trace 1 call at a time, so if many resources are being traced, it isnot predictable which call will be traced. The better way to obtain this information is to use the CDRTrace feature.NATEnable if the Resource is on a private LAN address, behind a NATing firewall. Disable if not. Indicatewhether or not the gateway is located behind NAT (network address translation). When accessing anetwork through a firewall, address translation can be necessary. To indicate address translation isrequired, select Enable. Select Disable to indicate that no network address translation is necessary. Thisfeature requires the Firewall to be configured to send to the SIP (UDP 5060) messages to the specifiedgateway. Otherwise traffic we send to the gateway may not reach it. To avoid this requirement, thegateway behind the NAT would have to register on an Access or Dynamic Peering Service Port. Thefunction of this setting is to ignore the provided SDP and to auto-learn the ports being used on each call.Allow Direct MediaUse this field to enable and disable a direct payload path from gateway to gateway that effectivelybypasses the VSXi. Select Yes to enable; No to disable direct media support. In order for the media tobypass the VSXi, this setting has to be set to Yes on both call legs. If one call leg is set to Yes and theother to No, the media is brought back through the VSXi. For greater control of this parameter you canset up several TIDs that point to the same resource, but with different media-handling characteristics. o Registrar TabThis configuration is only used when registering directly to the VSXi. If a feature server is supplyingregistrar functions, this configuration is not required in your VSXi since the feature server will containthe User information. (Figure 4.23) 50 Page
  52. 52. Figure 4.23: Registrar o Resource Block List TabUnder this tab, you can input digit patterns you wish to block with respect to certain resources. This isuseful for dealing with temporary outages that may not require full route table updates. You can enterthe specific country code in the digit match and the outbound TIDs you wish to exclude from the routelookups. You can also assign this to only specific inbound TIDs, not all calls. It can also be used toprevent calls from looping out to the same carrier the call arrived on. o SIP Profile List TabModify, Add or load defaults to this profile list. Choices are Load B2BUA Defaults or Proxy Defaults.Press Add to modify an individual field. The SIP profile allows you to customize the handling of SIPparameters. These can be assigned to the individual TIDs so that different TIDs are able to handle theSIP treatment uniquely. Each Profile allows customization for a particular type of device. A featureserver may require different settings than a remote gateway for instance. The Outbound Treatmentcontrols the outbound messaging for a TID. This means the settings are applied as the call is being sentto the TID using that profile. (Figure 4.24) 51 Page
  53. 53. Figure 4.24: SIP Profile  Major Tab: RoutesA route is designated by an Alias, Digit Match, and the Route Table it applies to. (Figure 4.25 and 4.26) 52 Page
  54. 54. Figure 4.25: RoutesRoutes Add 53 Figure 4.26: Route add Page

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