What Is Skype VoIP, And Who Is It For?
Skype VoIP is a platform that uses IP (internet protocol) to transmit voice packets. With this technology, users can make calls over the internet using any network that uses IP to transmit voice data. Voice signals are digitized, compressed, and converted into IP packets and then sent using IP networks. Signaling protocols locate the users, negotiate between users, create a connection and tear connection between users.
The advantages of Skype VoIP over Public Switched Telephone Network (PSTN) are Free calls if you’re on the same service, cheaper calls if you’re on different services, instant messaging, video calls, and integration with data transfer services for improved sound quality. In addition, the collaborative study will fundamentally assess the voice-over-internet protocol (VOIP) security threats, issues, and difficulties in communication over the system.
In this report, we talk about different VoIP security threats and possible ways to handle the threats in VoIP applications. There are some security issues, and threats have been discussed in detail. The countermeasure to mitigate these threats is also examined. General security services and mechanisms to protect VOIP from threats have been discussed in the paper. Different methods and channels, along with multiple layers to be protected to safeguard VOIP, are essential.
Basic Skype VoIP Architecture
The Skype VoIP architecture consists of four main components: Session Initiation Protocol (SIP), RTP, RTCP, and Real-time Transport Protocol (RTP). Every VOIP manufacturer has designed its architecture based on the propriety architecture and the protocols. As a result, the protocols and hardware change, but the basic functionality of the components remains the same in every architecture.
The VOIP of consists of four essential components:
Signaling Gateway Controller
The SGP is responsible for call control, which handles all the signaling information between the PSTN and VoIP networks. This includes routing calls from one network to another and handling any signaling errors that might occur. The SGP is also responsible for providing quality-of-service guarantees to callers, which means that it must ensure that calls are delivered promptly and without interruption.
- The Signaling Gateway Controller (SGC) provides bandwidth management control for multimedia sessions. The SGC allows admission control by admitting new sessions only when the bandwidth is available in the channel. The session is denied if the bandwidth is unavailable in the channel.
- The Bandwidth Policing mechanism is used in SGC to provide appropriate quality of service for multimedia sessions by instructing the media gateway to watch real-time transport protocol flow and applying policies according to rate limits.
- Call record details are another feature of SGC; it records calls companies use to charge their customers with more information on these calls.
- The Signal Gateway Controller (SGC) is a device that acts as a gateway between circuit-switched networks and packet-switched networks. The SGC supports media protocols MGCP or Megaco; both the MGCP and Megaco protocol are used to convert the signals from the circuit switch network to the packet switch network. The other work SGC does is to create, terminate, and maintain calls between multiple endpoints.
- The SGC is responsible for creating, terminating, and maintaining multimedia calls between endpoints using SIP protocols which are pure IP signaling protocols used for multimedia transport, bandwidth control, and session management.
A media gateway is a device that enables voice and data transmissions between networks. You can also use it to connect two communication platforms, such as PSTN (Public Switched Telephone Network) and VOIP (Voice Over IP).
The Media Gateway’s different roles in Skype VoIP architecture include:
- MGCP or MEGACO is both master-slave protocols used for call control. The control between the ports is maintained through these protocols.
- T1/E1 trunk support sends the voice data between the VOIP and PSTN networks.
- The media gateway supports different compression algorithms for fulfilling the requirements forwarded to the media gateway from the signaling gateway controller.
When utilizing VOIP, media servers add extra capabilities to the platform. Some of the extra features that you may introduce with media servers are:
- Voicemail (voicemail) is the feature of a telephone system that allows a caller to leave a recorded message for the called party. Businesses and other organizations typically utilize it to provide service when the called party is absent.
- Call processing tones and special announcements are the audible tones used to define the line’s status or equipment. Voice-activated dialing is a feature in VOIP which enables the user to dial a number using the voice.
- Voicemail to email transmission is the service in VOIP that enables the transmission of voicemail as an attachment in an email.
- Interactive voice response is a feature in VOIP that can interact with the user and gather information such as service activation, credit the plans, change plans, and route calls to appropriate recipients.
The application server is a component used in Skype VoIP architecture value-added services. For example, customer-specific services such as specific data plan with specific bandwidth explicitly designed for an individual customer is carried out using an application server. The other features that can be added using the application server are:
- The application server is a software platform that you can use to add advanced features to the VOIP architecture. It supports the Private Dial-In plan, which allows a number to be charged very low compared to the other numbers.
- The essential services such as forwarding the call to some other number, call waiting, and call busy are also functionalities that can be added using an application server to VOIP architecture.
- Advanced features like call authorization using the Pin and follows me plans can also add using an application server.
- The application server also maintains the call record details based on the requirement.
Skype VoIP Protocols Explained In Detail
Skype is a protocol that runs on top of TCP/IP. It’s a bit different than traditional VoIP systems because it consists of two types of nodes: ordinary nodes, which are used for audio calls and messaging, and super nodes (SN). Supernodes are the hosts situated on the edge of the Skype network.
To be successfully logged in, an ordinary node must register with the Skype login server and connect with a super node. For an ordinary node to become a super node, it must have a public IP address with adequate processing power, memory, and network bandwidth.
The Skype login server acts as a central hub for all the users and their requests. The requests are first routed through it before other nodes in the network handle them. It also acts as an intermediary between skype users and other systems such as applications, devices, and websites that use Skype API.
Essential Components Of Skype Protocols:
The Skype VoIP client opens a port on your machine to allow UDP and TCP connections. The client chooses a random port during installation. Although there is no standard port for UDP and TCP traffic, the client reserves port HTTP and HTTPS ports for listening to TCP traffic.
- Host Cache
The host cache or HC holds the catalog of all the super node IP addresses and port pairs that the skype client holds and updates. It has been considered the most critical function for working with Skype. The host cache must hold at least one valid IP address and port number entry.
- Buddy List
Skype digitally signs, encrypts, and stores the list of its trusted nodes or buddy list in the Windows registry database. The Skype client stores this list locally on each machine on which the application is run. In addition, the client repopulates the buddy list locally if the application is run on a new machine.
All Skype-based communication is encrypted using 256-bit ciphers. This process means that the data transmitted between clients and servers is encrypted, as well as the audio and video channels.
When you are connected to a Skype chat service, your client will use TLS security to encrypt all messages sent between your computer and the chat service installed on the server in the cloud. Using RSA certificates, the user’s public keys are certified by the Skype server at the login scenario.
Skype is currently making the internet more accessible to many people. As a result, more and more people are enjoying the benefits of this innovative VOIP system. You can use Skype itself both on personal computers as well as mobile phones. Microsoft provides various features to ensure organizations can choose the best Skype VoIP solution for their set-up and organization. However, if organizations are looking for a professional and reliable telephony provider, EPC Group can perfectly cater to their requirements. They can even provide a mix of both solutions to take advantage of Microsoft Teams features and communication channels.
Sas Chatterjee is a Senior Architect with EPC Group. His focus lies in making sure that the execution of each engagement is delivered in a forward compatible, best practices manner. Sas is an extremely devoted professional and takes each project he is assigned very seriously. During the project execution phase, Sas invests the time needed with his clients to gain a full understanding of their requirements and develops a roadmap for achieving their desired end goal.