Premises Installation Equipment: Fill & Download for Free

The satellite system is essentially a 3-part network, not just the spacecrafts in orbit. There's 1) the earth stations, 2) spacecrafts in orbit, 3) the customer premises equipment (CPE). Satellite broadband works through the transmission of signals between the customer premises equipment and the earth stations, via the satellite.The earth stations are the equivalent of mobile base stations. At these sites, there are multiple large dishes or antennas that are pointed to the satellites. These enable the uplink between the ground and the satellite. The dishes are supported by active electronics on the ground, which aggregate data into the terrestrial (backhaul) network. Earth stations are located in remote areas where there is very little population to minimise interference.The satellites in orbit receive the signals from the earth stations and send it on to all the premises covered. It covers a large number of premises by casting beams over a set area. Satellites can be configured with many small spot beams, each covering a different geographic area (http://www.wildblue.cc/wb1.jpg) or a single large broad beam covering an entire nation. A single satellite can have up to 50-60 spot beams. Multiple satellites can be designed with the same beam configuration (i.e. overlapping) to increase the total bandwidth or cover non-overlapping areas. A single satellite can have between 30-100 Gbps of throughput, depending on its size.Any premises that lies in the range of a beam will be able to receive satellite broadband services if it has the right customer premises equipment. A technician installs the external dish and points it at the right angle to communicate with a satellite. The antenna is then wired to an indoor modem.Rough numbers on bandwidth - not sure about the USA but in Australia, Telstra offers up to 800 kbps plans at A$500/month. The entry level service is 256 kbps at A$100/month. Government provides a subsidy to satellite subscribers through the Australian Broadband Guarantee.

Twilio is a company founded by developers for developers! Twilio provides a software-based platform which enables customers to easily add voice, messaging and video to their apps. Twilio is NOT selling a final product that can be consumed by an end-user (e.g., prepackaged software to solve their business problems), but rather is providing developers/customers with the prefabricated building blocks (i.e., APIs) needed to build any communication-based functionality they desire right into their application. By just virtually buying a phone number and swiping their credit card for potential per minute usage, developers can build contextually relevant communications by preventing their own customers’ users from leaving the application when they need to interact with someone (think why WeChat is popular in China). On this side of the world, popular use cases include ridesharing apps enabling anonymous communication between passengers and drivers and e-commerce companies sending automated delivery notifications or promotional messages.So how does it work?A developer signs up for Twilio, chooses a local virtual number (e.g., with 415 area-code for San Francisco, 212 for New York City) to send and receive voice, or SMS messages. The developer then maps the virtual number to a ‘request’ URL (the application’s URL, which Twilio would request from the developer’s application server when receiving a voice call on behalf of the developer/customer). The URL, or Uniform Resource Locator, created by the application developer, would describe to Twilio how to control the content of phone calls.The developer defines a set of business rules, or instructions to handling incoming and outgoing calls for each customer cohort. These instructions include: 1) Say - inform the customer that his or her order has been processed, or play a prerecorded sound file (message or music); 2) Gather - collect information from the caller; 3) Record - record the call; 4) Reject - hang up; and 5)Dial - dial this specific rep number to forward the customer to, or set up a conference call. In essence, on one end Twilio ‘ingests’ the phone call or the message flow, while on the other, it provides APIs for developers who, in turn, instructs Twilio on how to handle the incoming or outgoing phone calls. An API, or Application Programming Interface is a prefabricated block of software code that performs basic, reusable functions—e.g., displaying text on a computer screen, enabling inter-app communication—to allow developers to simply focus on building value-add, user facing applications.The call gets ‘load-balanced’ to a number of nodes/servers in Twilio’s cluster (managed by AWS). Each group of machines/servers perform specific roles—for example, some roles might be CPU-bound, a task is determined by the speed of the central processor, while others are memory-bound, determined by the amount of memory required to hold data—allowing Twilio to scale each group of machines independently of one another. The call comes in, gets load-balanced to a ‘Voice’ node cluster (i.e., a group of servers dedicated to voice function) to be answered by Asterisk, an open source PBX software platform.In general, a PBX allows telephone users to set up circuit-switched calls to other users in the same company (without toll charges) or to connect with users of the public telephone network. Effectively, PBXs shift some of the switching system out of the telephone company’s central office ‘CO’ and into the customer’s premise. The switch permits direct inward dialing ‘DID’ to a specific extension. Twilio stores the ‘DID’ numbers in its database; on the other end, the developer/customer would associate her or his assigned ‘DID’ number to a specific URL of a web application. As the call comes in, Twilio would make an HTTP POST request to that URL. In turn, Twilio would receive from the application server the XML instructions on how to handle the call.How does a phone call work? Historically, the telephone network has been a hierarchy of smaller networks performing different switching functions. In the US, switches were organized into classes from 1 to 5, Class 5 being assigned to the End Office or Local Exchange ‘LEX’. Classes 1 to 4 were Toll Centers or Transit Nodes ‘TN’. For a particular country, this network hierarchy would also be connected to international networks through a higher switch called International Gateway. Even if the modern structure has flattened, removing the Class 1 to 3 switches from the hierarchy, Class 4 and Class 5 designations remain in use today.The core of a telephone network is made of switches—TN and LEX. The primary function of these switches is to establish connections between telephones and the transmission equipment used to carry the voice call. When a called is placed, the switch detects that the receiver has been lifted, provides a dial tone and collects the called number. It then translates the dialled digits into a destination path by finding and reserving an idle circuit—if not it returns a busy-line tone. When the connection is established it sends a ringback tone to the caller until the destination picks up the phone. Then the connection is established by linking all the connections previously reserved in the transmission network and in the end terminates the call by releasing network resources reserved for the call.Typically, residential customers are connected to switches through an Access Node ‘AN’ or a Digital Loop Carrier ‘DLC’. This equipment converts the incoming voice into the appropriate digital format and also performs some basic call-processing functions. DLCs are access solutions that enhance the reach of the switch for remote customers. They do so by multiplexing many customer lines into just a few trunk lines that return to the switch.How does an internet connection work? The main historical way to access the Internet was a dial-up connection or narrowband access. Narrowband access uses the phone line to the Local Exchange, which is then connected to the data network through a Narrowband Access Switch ‘NAS’. The data network is typically a set of IP and/or ATM switches. The most widespread broadband internet access today is DSL. It re-uses the telephone copper wire up to Digital Subscriber Line access Multiplexer ‘DSLAM’ that directs the voice traffic onto the PSTN and the data traffic to the data network through a Broadband Access Router ‘BAS’. The PSTN, or the Public Service Telephone Network is a circuit-switched network that sets up dedicated voice circuits across a network of copper and fiber optic cabling.How do converged networks handle calls? There are many different ways to converge networks. In general, Network convergence requires the addition of two types of network elements: Media Gateways (MG) and Softswitches. Media gateways act as the physical bridge between the two networks. Their main purpose is to convert the TDM voice packet into data packets—generally IP or ATM. The softswitch acts as a media gateway controller by extending the voice network signalling ‘SS7’ into the data domain.When a phone call is placed on such a converged network, both the voice connection and the signalling are routed to the data network through a media gateway. The softswitch is a central database ensuring that Media Gateways properly communicate between each other.How does Twilio Fit in? To properly understand the inner workings of the Twilio platform, it is important to understand how basic web browsing work.What happens when you type a URL into a web browser. When you type aws.amazon.com for example, into a web browser, a behind-the-scenes process occurs translating the domain name (aws.amazon.com) into the IP address (54.239.31.69). The IP address is sent to the web browser so the user can be connected to the website. This process, called resolution, relies on a global network of name servers. The browser is a program that requests information from a web application server, which in turn finds the requested information and sends it back to the user’s client (desktop, mobile).A web server is essentially a software program that sits in the middle tier of an ‘N-Tier’ computing environment between client-side environments (web browser), and server applications. Application servers allows developers to focus on creating specific business logic objects without having to code at the system level. The primary role of the application server is to access enterprise servers (e.g., AWS’ servers) for business logic and data, serving application objects to clients in the form of static or dynamic HTML pages, JavaScript, etc.Clients and servers communicate using Hypertext Transfer Protocol Language (HTTP). IP addresses are what machines on the Internet use to identify one another. It is a 32-bit address defined by Internet Protocol, or IP that uniquely identifies each computer on the Internet. Every computer, or machine on the Internet has a unique IP address. If a machine does not have an IP address, it is not really on the Internet. When searching for frequently visited Websites (e.g., aws.amazon, Facebook, Google, etc.), the recursive name server already has the information cached in its memory and passes the IP address on to the user’s computer browser without having to take any further steps.The ‘conversation’ taking place between the browsers and the local recursive name server is analogous to the following dialogue:User’s computer: “What is the IP address of the URL Amazon Web Services (AWS)recursive name server: “I already know that. The IP address is 54.239.31.69.” The local recursive name server then delivers the address to the end user’s browser.Under ‘the cover’, a ‘load balancer’ differentiates between a request to a main page and another to its extension (e.g., www.amazon.com and today’s deal extension www.amazon.com/gp/goldbox ),and sends the request for the web page of today’s deals to a group of servers that is optimized for that task. A load balancer goes one step further by allowing traffic routing decisions to be based on the so-called HTTP header (HTTP, or Hypertext Transfer Protocol is an application-level protocol that is the foundation for the World Wide Web). A load balancer’s controller not only examines the URL, but also information such as cookies, client source address, etcTwilio Client SDK enables VoIP communications. Twilio Client enables developers to integrate Voice Over IP, or VOIP into their applications.To illustrate, let’s examine a scenario whereby hypothetical InsuranceCo were to roll out Salesforce CRM Call Center mobile application through which is provisioned several customer service numbers (local numbers, international, 1-800 etc.) provided by Twilio. InsuranceCo buys a virtual 1-800 number from Twilio. InsuranceCo’s customer calls the 1-800 number. Salesforce has set up workflows, on behalf of InsuranceCo, for how to deal with inbound calls. Twilio Client SDK enables VoIP communications within the Salesforce application. The customer’s carrier (e.g, Verizon) routes the call to a Competitive Local Exchange Carrier ‘CLEC’ with whom Twilio has contracted (e.g., Level 3) . The call is routed through AWS to Twilio, triggering an HTTP request to Twilio’s call API to initiate the call.On the other end, Twilio 1) looks up the voice URI (Uniform Resource Identifiers) of the Salesforce Service Cloud application, and 2) provides Salesforce’s web server +information about the caller (e.g., originating and dialed phone numbers, date and time of the call, geographic location of the caller, etc). Twilio would also request instructions from the Salesforce application server on how to handle the call. The process is quite similar to how a browser makes a request to a web server discussed above.Once the Salesforce application receives the caller’s information (i.e., the call is logged into Salesforce). Processing function follows, involving a caller/customer database lookup to map customer’s caller ID to her or his historically stored information (e.g., caller's support ticket history). Caller’s information are then injected into the Salesforce application workflow. In this example, InsuranceCo.’s marketing database (maintained by Salesforce) has been capturing data about the customer’s behavior from various sales and communications channels.Twilio’s TaskRouter. Salesforce’s application server then sends to Twilio’s server (which is hosted on AWS) XML instructions on how to handle that specific phone call (e.g., since the analytical engine determined the caller is a high value distributor/customer, please forward immediately the call to highly skilled call-center rep A). An HTTP request is sent to a virtual call router’s API (i.e., Twilio’s TaskRouter's API). The call router, or TaskRouter authenticates, or looks up the account holder’s unique ID, matches the caller to the intended rep, then sends an HTTP request to Twilio’s server to initiate the phone call between the caller and the rep.The existing data on the caller/customer is enhanced by the overlay of the real-time contextual communication data elements; i.e., putting the customer call in context of the Salesforce’ customer service application, providing the call center agent real-time intelligence, needed for improved customer profiling. The agent can instantly access and view prior communications, use real-time analytics identify the personalized best next-action for the customer (e.g., promote a special offer, provide a service alert), and through, for example, Pitney Bowes’s content-creation platform, automatically integrates incoming data, compose new offer and send it to the customer through the relevant channel.What is Voice over IP ‘VoIP’? Voice over IP (VoIP) is a technology that involves sending telephone calls over data networks, such as the Internet, rather than the traditional Public Switched Telephone Network ‘PSTN’. Traditional phone calls across the PSTN use a dedicated circuit that transfers calls as uncut streams, allowing no other information on the circuit regardless of available bandwidth. Conversely, IP networks transfer data more efficiently in packets that get reassembled on the receiving end rather than using a dedicated circuit. For a VoIP call, audio is first converted from an analog signal to a digital signal through a codec, separated into discrete packets, sent across an IP-based network, put back into order at the termination point, and then converted back to an analog signal to create audio that the end user can recognize. There are three major VoIP protocol standards—namely, H.323, SIP, and MGCP.The Session Initiated Protocol, or SIP In March 1999, the Internet Engineering Task Force ‘IETF’ introduced Session Initiation Protocol ‘SIP’, which was designed to support quicker call set-up times and enhanced Web capabilities. SIP does not require gateways to maintain all call information during the life of the call and, as a result, it is less demanding of servers and is more scalable.SIP focuses on session initiation, modification and termination, and leaves the session and connection details to be negotiated by the end points. SIP uses a simple text command structure with HTTP syntax and URL addressing. Thus, SIP is well suited for any Internet- and Web-based applications.SIP allows direct communication between clients via a peer-to-peer ‘P2P’ connection using only IP addresses, and in most cases a SIP proxy will query a Domain Name Server ‘DNS’ to resolve a domain name into an IP address of record. An important characteristic of the SIP protocol is that it allows endpoints to establish a peer-to-peer connection and communicate directly. This is in contrast with other protocols that require resources on the network to control communications between end points.How does SIP “Work”? To initiate a phone call, the SIP phone issues an INVITE containing the caller’s IP address and the type of media requested (e.g., voice). From there, the SIP servers (Proxy and Redirect server) send back the SIP-URI where the called party can be reached. In this respect, the SIP servers somewhat resembles the DNS, or Domain Name Server servers on the Internet.What is the Domain Name System ‘DNS’? The Domain Name System (DNS), a global, distributed database infrastructure, is part of the fabric that holds together the Internet—performing the simple, straightforward function of mapping URLs (Uniform Resource Locator) to IP (Internet Protocol) addresses. Every Web server on the Internet has one or more unique IP addresses, represented as four sets of numbers, called octets, separated by periods (e.g.,140.211.169.9). The DNS enables people to use domain names (e.g., www.opendaylight.org ), which are much simpler to remember, to identify Web servers as opposed to IP addresses (e.g., 140.211.169.9). Each time a user enters a domain name into a computer’s Web browser, the DNS translates the user-friendly URL into the IP address needed to locate the appropriate Web server.SIP uses the above request and response method to establish communication among the various network components and ultimately enable a multimedia conference between two users. Users are identified by globally reachable unique addresses called URIs. URIs use the same format as email addresses to describe IP service points (e.g., [email protected]). Users register their assigned URIs with the registrar server, which provides this information to the location server upon request. Users can have multiple URIs with different service providers that point to the same device, but they can also be reached with traditional telephone numbers. Calls using these traditional numbering schemes are translated into SIP URIs using the ENUM method.E.164 Number Mapping ‘ENUM’: Telephone Numbers on the Internet ENUM, or E.164 Number to URI Mapping translates telephone numbers into Internet addresses. ENUM is a protocol that merges the Public Switched Telephone Network ‘PSTN’ and the Internet—mapping complete international telephone numbers to URIs. Since the SIP protocol is IP based, it provides users (and applications) globally reachable addresses called URIs (Uniform Resource Identifiers). URIs use the same format as email addresses to describe IP service points (e.g., tel: [email protected], mms: [email protected], etc.) and can be reached with traditional telephone numbers. Calls using these traditional numbering schemes can be translated into SIP URIs using the ENUM methodology.To put ENUM into context with the aforementioned technologies, SIP performs the initiation of interactive communications sessions between users, as well as termination of those communications and modifications to those sessions. SIP is one protocol that may be used by ENUM to send out initiation attempts to multiple locations in order to find the user who is receiving a call. By placing telephone numbers into the DNS, ENUM can facilitate a range of applications including addressing for fax machines, email, instant messaging, etc.What Value Does ENUM Add? ENUM enables users to access Internet services from Internet enabled phones, ordinary phones that are connected to Internet gateways or proxy servers and/or other Internet devices that may have inputs limited to a numeric keypad.ENUM also provides users with greater control over communications, including allowing users to indicate their preferences for receiving communications. For example, a user can indicate a preference for voice mail messages over live calls during certain times, or may specify a call forwarding location.ENUM allows an end user to reach an IP device by dialing a telephone number rather than entering a URI. A traditional number is entered into the calling device, and the number is then transformed into a fully qualified address by an application or a device that supports ENUM.A developer’s customer dials a Twilio provisioned virtual phone number. The SIP proxy queries the ENUM DNS server to resolve the fully qualified domain name into a URI. The SIP proxy will then query a DNS server to determine the IP address to send the invite. To illustrate, let’s walk through a scenario whereby End User A dials a Twilio’s business customer, or User B:User A dials User B’s phone: +1-646-470-8021.Internet Gateway converts number to a domain name and queries VoIP local recursive name server: 1.2.0.8.0.7.4.6.4.6.1.e164.arpa. By using ENUM, e.164 numbers can be used in DNS by transforming the phone number into a hostname. This is simply done by reversing the numbers, separating each digit by a dot and then adding the e164.arpa suffix.13 For example, the number +1-646-470-8021 would be transformed to the fully qualified domain name 1.2.0.8.0.7.4.6.4.6.1.e164.arpa.Local recursive name server: “I don’t know that address, but I’ll check with a name server that does. Hold on for a millisecond.”e164.arpa Tier 0 server: “Here are the addresses for the authoritative name servers for the CC1 1.e164.arpa domain.”domain.com name server: “The URI for 1.2.0.8.0.7.4.6.4.6.1.e164.arpa is [email protected].”User A’s telephone contacts User B’s (Twilio’s customer’s) telephone at returned IP Address. The local name server launches a query to the DNS, which responds with the IP address (e.g.,108.231.245.239) of the local proxy server associated with [email protected]. The SIP proxy server in User A’s network contacts the SIP proxy server in User B’s customer network, and the proxy server in User B’s network then contacts User B’s destination SIP IP phone.When the called agent receives the INVITE request, it determines if it can accept the call. If it can, it will ring the phone and then send back a response to let the calling agent know that it is ringing. When the phone is answered, User B’s, or Twilio’s business customer’s phone sends an OK response back to the calling agent along with its media capabilities. Both agents agree on a media channel, and User A’s phone sends an ACK to User B’s phone.After responses and acknowledgments are exchanged, an RTP ‘Real-time Transport Protocol’ session is established between SIP IP phones of Users A and B, or the end-user and Twilio’s business customer.Traditional Voice and the Public Service Telephone Network ‘PSTN’. The voice telephone systems are referred to as the PSTN ‘Public-Switched Telephone Network or POTS ‘Plain Old Telephone System’. PSTN is a circuit-switched network that sets up dedicated voice circuits across a network of copper and fiber optic cabling. The structure of the traditional telecommunications industry varies by country and depends on the nature of the regulatory environment.In the United States, the industry has been pushed into a competitive model consisting of a variety of participants, primarily oriented around local exchange carriers ‘LECs’ that provide last-mile connection for consumers and businesses within specific geographies ‘LATAs’, and Interexchange Carriers ‘IXCs’ that provide long-distance services. A more graduated categorization includes ILECs ‘Incumbent Local Exchange Carriers such as SBC/Verizon), CLECs ‘Competitive Local Exchange Carriers’ (e.g., Level 3), IXCs (e.g., MCI/Verizon and AT&T), and ISPs ‘Internet Service Providers such as Earthlink and Prodigy/AT&T.Twilio interconnects to the so-called Tier 1 carriers - carriers that own or control sufficient portions of their underlying network infrastructure;e.g., Verizon, Level 3 - to provide the PSTN integration. It uses SIP (described above) origination and termination to ‘talk’ to originate and terminate calls on the PSTN.Traditional PBX. As discussed above, a PBX is a telephone switch located on the premises of a company. Traditional PBXs were typically hardware-based solutions that ‘sat’ inside customer premises ‘CPE’, providing businesses with the benefits of direct dialing, call forwarding, and a variety of enhanced services. Put another way, PBXs were originally designed for businesses as a cost-effective alternative to the provisioning of individual lines to each end-user from the phone company’s central office. The PBX is like a mini-CO, owned and operated by the corporation itself. In this respect, traditional PBXs reduced both line provisioning costs for the corporation and telecom services expenses associated with intra-office calls.Mostly Proprietary. Nonetheless, because PBXs are highly proprietary systems, enterprises have had to rely heavily on the PBX vendor to deploy or integrate any new applications. In a traditional PBX, there is a proprietary operating system running on a computer processor in a proprietary chassis. The applications are also proprietary, running on the same or separate processors. The interfaces—trunk cards and line cards— are also proprietary. In short, the traditional PBX is like a black box, with the vendor controlling virtually everything—the addition or adjustment of applications generally needs to be made by the PBX vendor. The proprietary nature of the technology is often predicated by its complexity. In fact, a high-end legacy PBX usually incorporates about 5 million lines of code.Limited Scalability. One of the most significant limitation of legacy PBX systems may be scalability. PBXs are typically designed for a specific number of users, and once the enterprise expands beyond that specific capacity, a new and bigger PBX needs to be installed. Sometimes, small businesses have to purchase a higher-end PBX than they need in case of possible future expansion, resulting in a particularly inefficient use of capital. It is also problematic to connect PBXs across multiple sites, and the signaling between PBXs is proprietary. Another key problem is that handsets which customers may have purchased for the lower-end PBXs often do not work with higher-end PBXs. As a result, customer upgrades to a higher-end PBX system often necessitate the additional cost of purchasing new handsets.Asterisk and IP-PBXs. Enterprise networking focuses on three primary goals: 1) Scalability, 2) Controlling the cost of communication through the most efficient use of technology and carrier services, and 3) Improving the productivity and performance of workers by distributing information to support their activities. With the advent of the Internet, IP PBX systems were introduced and allowed for phone calls to be placed over IP-based, rather than over TDM-based networks. In such an IP environment, distributed communications servers ‘IP-PBXs’ provide scalability and redundancy by sharing and quickly reconfiguring resources in the event of individual server failure. This redundancy and distributed processing is only feasible because the architecture separates the voice traffic from the PBX, leaving only call signaling and processing responsibilities to the PBX; hence, enabling independent scalability.Asterisk, on the other hand, is commonly used open source PBX software platform, developed in 1999 by a company called Linux Support Systems that later changed its name to Digium. The development of Asterisk is predicated on the idea that modularity, or separating a PBX system into interconnecting components—akin to a boxful of LEGO bricks—would enhance scalability. An Asterisk based IP-PBX is essentially a x86 communications server, running on Linux.Hosted PBX With Hosted PBX, PBX functionality is delivered as a service over the carrier’s network. Enterprise customers typically pay for the service under a leasing arrangement. Rather than having a PBX system located on the enterprise’s facilities, those functions are located in the carrier’s network and delivered over IP-based trunks to the enterprise.SIP Trunking involves the direct IP connection of a SIP-enabled IP-PBX and SIP-compliant VoIP service provider. It is an IP-based service provided by telecom operators (and Twilio) to connect an enterprise’s PBX with the service provider’s network. Put another way, SIP Trunking is an IP connectivity consisting of a single pipe which connects a service provider’s network to an enterprise IP PBX. As discussed earlier, SIP, or Session Initiation Protocol, is the protocol used to set up IP-based sessions between network endpoints such as end-user devices or servers. SIP trunks allow operators to provision VoIP voice sessions. Enterprises benefit from SIP trunking as it consolidates their voice and data networks, replacing premise-based connectivity, and thus reducing overall costs. In the past, enterprises had to connect bundles of physical wires—PSTN lines—to a business—PSTN endpoint. A SIP trunk eliminates the PSTN lines—reducing the number of SIP connections per port—and other associated equipment such as PSTN gateways.Thus, SIP Trunking offers a number of advantages over traditional TDM-based connectivity. First, SIP Trunking allows the enterprise to reduce its telecommunications costs. While many enterprises already save on the cost of voice calls between their sites by implementing IP-based PBX systems and using intra-corporation VoIP calling. Using SIP Trunking, enterprises can further expand their ROI by extending VoIP outside of the corporate LAN.The savings comes from: 1) Getting rid of traditional analog/POTS, ISDN BRI ‘Basic Rate Interface’, ISDN PRI ‘Primary Rate Interface’, or T1/T3 subscriptions; 2) More optimal use of SIP trunk bandwidth as both voice and data services can be delivered over the same connection; 3) Greater flexibility in purchasing voice capacity as enterprises don’t have to purchase lines in groups of 24 T1 or 30 E1 lines; 4) Flexible routing of calls to preferred carriers —long distance calls can be made for the cost of local calls; and 5) Lower operating costs of IP-PBX systems vis-à-vis traditional TDM-based PBXs.For most enterprises, the desire to save money is the primary force driving adoption of SIP Trunking. Least cost routing is an interesting example. Enterprises can utilize SIP trunks from multiple service providers and proactively route specific calls to certain carriers based on country codes—operators often charge different international rates based on availability, time zone differences, and geography.Moving beyond these types of cost-reduction initiatives, SIP Trunking enables a host of additional capabilities that enterprises can benefit from. As discussed earlier, the SIP protocol itself was designed to initiate all types of real-time communications over IP networks — not just voice. Today, enterprises are taking advantage of not only the significant cost savings afforded from SIP Trunking, but also the ability to improve enterprise productivity through the deployment of Unified Communications applications.Where Twilio fits in: Elastic SIP Trunking.Companies such as Acme Packet (acquired by Oracle) use their Session Border Controllers SBCs as SIP firewalls. A session border controller is a piece of network equipment or a collection of functions that control real-time session traffic at the signal, call control, and packet layers as they cross a packet-to-packet network border between networks or between network segments. SBCs are typically located at the perimeter of disparate IP networks, such as between headquarter and branch offices, and/or between call centers and enterprise data centers. They provide network operators with ‘policy-based control’ over VoIP sessions, furnish basic protocol inter-working and defend the carrier backbone against a variety of attacks.SIP trunks are VoIP trunks from service providers that use SIP for call control and routing,enabling enterprises to create a single, pure, IP connection to a carrier’s core network. This can be viewed as an Enterprise IP-PBX that peers with the service provider core SIP proxy. “Twilio-SIP is for use in TwiMl type applications to terminate or originate a call from a known SIP endpoint or address. Elastic SIP Trunking is utilized when you have an existing application or appliance that needs to have origination and termination capabilities (think a PBX like Asterisk/Freeswitch) and Twilio will be that provider”.Programming Messaging.Twilio’s Messaging API enables developers to embed text-based communications in their applications. Using the same virtual number, our hypothetical InsuranceCo (discussed above in the voice section) can use the same both make and receive voice calls, and send and receive SMS.SMS would follow a similar pattern of voice--the flow from caller/sender to receiver--yet with the addition of an SMS aggregator. Say for example an Uber rider contacts a driver. The Uber application would use Twilio’s API to generate and initiate the SMS to the driver’s number. The SMS is routed through AWS to a CELC (e.g., Level 3). The CELC would then route the message to an SMS aggregator with whom Twilio has contracted (e.g., Syniverse). The SMS aggregator routes the SMS to the driver’s carrier (e.g., AT&T). The driver’s carrier then routes the SMS to the driver.The SMS Aggregator. An SMS aggregator such as Syniverse (acquired by The Carlyle Group)is essentially a ‘clearinghouse’ provider that facilitates wireless roaming between different carriers’ networks. As a third party intermediary, it plays a necessary role in a complex telecommunications environment characterized by different network architectures, signaling standards, billing record formats, and network protocols. The aggregator serves as the middle hub connected to all carrier partners, allowing each to roam on other’s network (assuming a roaming agreement is in place).Continuing on with the Uber example, when the SMS is sent by the rider, a detail record is created. This detail record contains basic information about the SMS (e.g., who is the sender, where they sent, the length of the message, the carrier that authorized the message). This record is then stored in one of the several formats. For GSM, the format is known as TAP, or Transferred Account Procedure (TAP) file, while for CDMA the format is known as CIBER.The data record must now be communicated to the right partners—this is where an SMS aggregator comes in. Syniverse, for example, receives the information in its data center, aggregates the data, and distributes the information to the right carriers. The company then calculates the net obligation of each carrier based on the information detailed by the data records.The Short Message Service Centre (SMSC). SMS includes a number of distinct features, which I have highlighted below. These are made possible as messages are sent via an SMSC. The SMSC mainly acts in a similar way to a router of messages. However, it also acts as an important interface with other parts of the network and other systems on that network.In general, one of the main features of an SMSC is the ability to store and forward messages. If the receiving device is switched off, the central system stores the message until it receives a signal that the device is now switched on, when it will then deliver the message. In addition, the functions of an SMSC can include providing the senderInterface with Other Network Elements. In addition to the store and forward features, the SMSC can provide an important interface to an operator’s other applications and act as the router between these. For example, the SMSC may interact with the pre-paid billing system, location servers, user profiles and platforms for other SMS based applications.

Disadvantages of Fiber Optic TransmissionThough fiber optic transmission brings lots of convenience, its disadvantages also cannot be ignored.Fragility: usually optical fiber cables are made of glass, which lends to they are more fragile than electrical wires. In addition, glass can be affected by various chemicals including hydrogen gas (a problem in underwater cables), making them need more cares when deployed underground.Difficult to Install: it’s not easy to splice fiber optic cable. And if you bend them too much, they will break. And fiber cable is highly susceptible to becoming cut or damaged during installation or construction activities. All these make it difficult to install.Attenuation & Dispersion: as transmission distance getting longer, light will be attenuated and dispersed, which requires extra optical components like EDFA to be added.Cost Is Higher Than Copper Cable: despite the fact that fiber optic installation costs are dropping by as much as 60% a year, installing fiber optic cabling is still relatively higher than copper cables. Because copper cable installation does not need extra care like fiber cables. However, optical fiber is still moving into the local loop, and through technologies such as FTTx (fiber to the home, premises, etc.) and PONs (passive optical networks), enabling subscriber and end user broadband access.Special Equipment Is Often Required: to ensure the quality of fiber optic transmission, some special equipment is needed. For example, equipment such as OTDR (optical time-domain reflectometry) is required and expensive, specialized optical test equipment such as optical probes and power meter are needed at most fiber endpoints to properly provide testing of optical fiber.