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Hello, let’s first define each of the two certificates.CompTIA Security+:The CompTIA Security+ exam will certify the successful candidate has the knowledge and skills required to install and configure systems to secure applications, networks, and devices; perform threat analysis and respond with appropriate mitigation techniques; participate in risk mitigation activities; and operate with an awareness of applicable policies, laws, and regulations. The successful candidate will perform these tasks to support the principles of confidentiality, integrity, and availability.This exam covers 7 general topics:Network SecurityCompliance and Operation SecurityThreats and VulnerabilitiesApplication, Data, and Host SecurityAccess ControlIdentity ManagementCryptographyRecommended ExperienceCompTIA Network+ and two years of experience in IT administration with a security focusExam Price:320.00 USD. You also must re-certify every three years.Exam Languages:English, Japanese and Portuguese (Simplified Chinese will be added in 2018)Security+ is government approvedCompTIA Security+ meets the ISO 17024 standard and is approved by U.S. Department of Defense to fulfill Directive 8570.01-M requirements. It is compliant with government regulations under the Federal Information Security Management Act (FISMA).Security+ provides substantial earnings potentialAccording to the Bureau of Labor Statistics, Security Specialists, Administrators and Managers earn over $86,000 per year. (I think this number isn’t accurate, it is exaggerated. You need much more than S+ certificate to get this number.)Systems Security Certified Practitioner “SSCP”:The SSCP shows you have the technical skills to implement, monitor and administer IT infrastructure using information security policies and procedures. You’re skilled in protecting the confidentiality, integrity and availability of data. You matter!This exam covers 7 general topics:Access Controls 16%Security Operations and Administration 17%Risk Identification, Monitoring and Analysis 12%Incident Response and Recovery 13%Cryptography 9%Network and Communications Security 16%Systems and Application Security 17%The SSCP is ideal for those working in roles such as:Network security engineerSystems/network administratorSecurity analystSystems engineerSecurity consultant/specialistSecurity administratorSystems/network analystDatabase administratorExam price:250.00 USD and you must pay a $65 annual maintenance fee. You also must recertify every three years by earning 60 Continuing Professional Education (CPE) credits.Security+ is government approved:SSCP certification is one of the US Department of Defense (DOD)-approved baseline certifications for both Level I and Level II Information Assurance Technical (IAT) certifications.Earning PotentialAccording to PayScale - Salary Comparison, Salary Survey, Search Wages, the average SCCP holder has between one to four years of experience in IT, and earns as much as $88,000 per year. The highest paid employees with the SCCP are those who also hold a master’s degree in a related field. (Again I think this number isn’t accurate, it is exaggerated. You need much more than SSCP certificate to get this number.)Security+ VS SSCP:Both are vendor neutral entry-level certification. As any entry-level certification both are too basic to have real skills but it’s essential stepping stone needed to enter the IT security field.Although SSCP exam passing score is lower, my opinion is that Security+ exam questions are more direct and less challenging. I think SSCP exam is a little bit more challenging to pass.Security+ is more popular so you can have much more studying materials options and in-class courses available.Security+ is more in-demand and more valuable. SSCP less commonly requested than the Security+ in terms of job requirementsVerdict:I believe CompTIA Security+ is a better way to inter the IT security field. Although the exam cost more compared to the SSCP but it’s more valuable.Sources:CompTIA Security+IT Security Certification | SSCP - Systems Security Certified Practitioner | (ISC)²8 Most Difficult IT Security CertificationsMaking Sense of Certifications: SSCP vs. Security+ - CybraryIf you are interested in CompTIA Security+ study materials and exam practicing I recommend www.AlphaPrep.netUpvote if you find this useful.Comment if you have more questions.Follow if you are interested to read more about computers and networks.Have a nice day :)Related Questions:Is it worthwhile to get CompTIA Security+ certification?What are the best books from a beginner to cyber security specialist? What languages or practice sites would you recommend?Should I pursue Security+ before CEH?How hard is CompTIA Security +?What should I do to be an expert in network security?>Study Groups Questions:CompTIA A+: You are ready to install a new Thunderbolt card. You plug the card into a PCIe x4 slot. What should you do with the cable and connector indicated in the figure?Network+: Refer to the exhibit. What command produced the output shown on your Windows workstation?Security+: Which of the following provides for the best application availability and can be easily expanded as an organization’s demand grows?CCNA: Refer to the diagram; assuming S1 is acting as a DHCP server, on which device and using what command would DHCP relay be correctly configured?CCNP: BGP’s choice of best path. Which of the following is least likely to be used by the configuration in this implementation plan?

CERTIFIED INFORMATION SYSTEMS SECURITY PROFESSIONAL (CISSP)The Certified Information Systems Security Professional (CISSP) is an information security certification for security analysts. It was created by the International Information Systems Security Certification Consortium (ISC). The certification was created to ensure professionals in computer security have standardized knowledge of the field. Earning a Certified Information Systems Security Professional certificate can help you have a successful career as a computer security professional.WHAT IS THE CISSP?The Certified Information Systems Security Professional (CISSP) exam is a six-hour exam consisting of 250 questions that certifies security professionals in ten different areas, of access control systems and methodology, business continuity planning and disaster recovery planning, physical security, operations, security, management practices, telecommunications and networking security. Other areas important to the CISSP certification are cryptography, security architecture application and systems development, law, investigation, and ethics.HOW TO BECOME CERTIFIEDTo become certified as a CISSP, you will need at least five years of full-time, paid work as a security analyst in two or more of the eight domains covered in the CISSP, such as cryptography and software development security There are experience wavers available for those with college degrees and additional credentials if these are approved by the (ISC). You also can become an Associate of the (ISC) and earn the CISSP when you meet your experience requirement. The next step is to prepare for and pass the exam. You will need to have a score a minimum of 700 out of 1000 points to pass the exam.After passing the exam, you will need to have an endorsement in subscribing to the (ISC) Code of Ethics and you will have to have endorsement from another (ISC) professional who can verify your professional experience requirements such as length of employment, professional reputation, and continuing education as a security analyst. Obtaining a professional endorsement from a current member is the reason it important to belong to professional organizations and to participate in professional seminars and events. These are useful career moves that are useful for networking with potential endorsers of your CISSP application.A CISSP certification requires an annual maintenance fee of $85 at the end of each certification year, and you must take the test every three years to remain a member in good standing with the certification. You must earn at least a minimum of 20 Continuing Professional Education (CPE) credits each year within the three-year certification cycle. You can re-certify by completing 40 annual CPEs and paying the annual maintenance fee. These classes can be taken at a university or through online courses dedicated to security issues.HOW TO PREPARE FOR THE CISSPSecurity professionals who study for the CISSP should be able to explain issues such as architecture and access control for protecting information system assets. In being able to explain these issues to clients and other stakeholders, the analyst must know how to assess the business or organization's current operations policies for incident response and make recommendations to those concerned for improvements to business or organization security. Knowing how to explain the importance of disaster recovery policies and demonstrate multiple and effective strategies to clients and stakeholders is a key skill tested in the CISSP. As part of the communication process, security analysts must compare and contrast different cryptographic protocols and be able to make recommendations based on this analysis of security needs. Creating systems of policies, standards, procedures, and guidelines with clients and stakeholders in mind should be the end goal of a CISSP analyst who earns certification.In terms of technical knowledge, CISSP analysts must demonstrate proficiency in a number of areas. Proficiency in network architecture and design, being able to implement network architecture to anticipate threats and best use given sometimes limited resources. This includes demonstrating clear understanding software security applications life cycle effectiveness. CISSP analysts also should have the ability to collect digital forensic evidence while maintaining the integrity of the evidence gathered. They also must demonstrate knowledge of physical security systems and how they add value to network security systems.WHY GET CISSP CERTIFICATION?A Certified Information Systems Security Professional (CISSP) is essential for security professionals to be successful in their careers. Many employers value the CISSP for its designation as a standard for security professionals. While the investments in time and money are substantial, the career rewards can be valuable as professionals with the CISSP are in demand.Burning Glass Technologies, a career site, reports that nearly one fourth of cyber security job postings in 2015 requested the CISSP. According to the (ISC), "certified information security professionals earn a worldwide average of 25 percent more than their non-certified counterparts." Being a CISSP professional can lead to higher pay and a more rapid advancement in the security analyst field. Security professional positions such as network security specialists, senior security engineers, information security manager, or chief security officers can all benefit from CISSP certification training.Search for: https://certsprepare.com/exam-dumps/CISSP/

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.