Voice over Internet Protocol, Component Functionality, and Related Processes

This paper describes the functionality of gateways and routers, IP PBX systems, and IP telephones and provides examples to demonstrate those concepts.  It discusses the characteristics of circuit-switched networks and packet-switched networks.  Also discussed is how various components, protocols, and services are used to make VoIP calls over traditional PSTN and other legacy networks.  This paper explains the role that various protocols fill in the process of delivering high-quality digital voice communications.  It describes the different voice-port connection modes available on Cisco routers.  The paper explains the session initiation protocol and how it’s applied during the VoIP process.   It analyzes the process to set up a VoIP call across a VoIP network using SS7, Softswitch, media gateways, and PSTN switches and provides examples to demonstrate those concepts.  It also highlights future trends with VoIP technologies and trending security risks and vulnerabilities.  

            Keywords:  voice over internet protocol, circuit-switched networks, packet-switched networks, interconnected networks, converged networks, media gateways, session initiation protocol, session initiation protocol, public switched telephone network, signaling system 7, differential services, softswitch, multimedia, sip trunk vulnerabilities, unified communications

Voice over Internet Protocol, Component Functionality, and Related Processes

Co-existing with traditional telephone lines and infrastructures is a technology that allows voice communications over the Internet.  Voice over Internet Protocol (VoIP), in its most basic form, allows users to place calls over the internet to other users who are also connected to the Internet (DeSantis, 2009).  Interconnected components and services allow users to receive and make VoIP calls to and from traditional circuit-switched networks that have been used since the late 1800s.  There were 47,900 telephones using these networks in the United States by the end of 1880, and in 1915 transcontinental phone service began.  

Circuit-switched networks select a path for each call and then dedicate the created circuit to that call (refer to figure 1).  Packet-switched networks break messages into small packets and then route the packets across many different paths and reassemble the packets at the destination.  This technique does require a dedicated connection, making better use of the network by allowing a higher number of simultaneous calls to be placed.  Today it is common practice to initiate a voice call using VoIP in combination with interconnected services, and infrastructures like public switched telephone networks (PSTNs).

This paper discusses how various components, protocols, and services are used to make VoIP calls over traditional PSTN and other legacy networks.  Also, it should be noted that gateways and specialized routers allow for calls initiated from within traditional networks to take advantage of Internet protocol telephony services and cellular networks.  Making a call from a traditional payphone to a cellular phone may be considered an example of that process.  It is assumed that the reader is familiar with the many acronyms listed in the appendix, as they are necessary to explain and analyze the concepts, functionality, and processes described in this paper.            

Voice over Internet Protocol Functionality, Gateways, and Routers

Voice over Internet Protocol (VoIP) networks utilize routers that support VoIP protocols like MGCP, SIP, and H.323.  Routers that are SIP-compliant allow the integration of data and voice, mobile, and web services on VoIP networks.  Multimedia transmissions are facilitated using H.323-compliant routers.  Routers within the network can serve as media gateways and are then used to perform circuit to packet converting tasks and call control; while communicating with media gateway controllers and must be MGCP compliant.  There are four main categories of gateways used in IP Telephony:

  • High-Density Media Gateways – Used at the core layer, can interconnect networks and can be used as a high-density call completion platform (see figure 3).  They are suited for migrating legacy infrastructures to VoIP services and provide PSTN trunking for call centers (see figure 4).
  •   Low-Density Media Gateways – Used at the access layer, used for lower density applications like the support of remote locations or for delivering hosted services.  They can provide any-to-any switching capabilities like high-density gateways while providing a more economical solution.
  •   ATM Media Gateway Switches – Provides interconnection between different protocols and physical connections while providing smooth transmission of streaming media content.  The gateways are H.323 compliant.
  • Next-Generation layer 2 and 3 Switches and Routers – Typically, these devices are used in high-performance networks to address the requirements of several devices while using a single platform.  Layer 3 switches can interconnect networks and/or connect directly to the Internet.

When routers are used as gateways, they should support the following:

  • Ip precedence tagging and processing or DiffServ,
  • voice packetization,
  • telephone signaling,
  • connection control,
  • and audio compression.

Routers can also be used as gateways for GPRS and GSM.  When used for connecting dissimilar networks the following tasks can be performed:

  • The execution of QoS buffering and queueing,
  • Identifying, translating, and routing voice packets,
  • and the proper and appropriate prioritizing of traffic.            

Voice Port Connection Modes on Cisco Routers

Connections modes on a Cisco router can be configured to specify the system used for signaling, define busy or no-answer responses, and select the system that will translate the DTMF digits.  Cisco routers allow for five connection modes that can be configured:

  1. Trunk – When properly configured in this mode a PSTN trunk is established when activated.  Trunk ports can be configured to stop activity when no voice traffic is present, a common bandwidth saving technique.
  2. Tie Line – Used to connect two PBX systems, simulating a legacy PSTN tie line.
  3. Switched – When in this mode the port must be configured with a peer to dial the intended number.  This mode does not support trunk connections.
  4. Private Line Automatic (PLAR) – This mode configures the port to automatically connect to a single private number.
  5. PLAR for Off-Premise Extensions (PLAR-OPX) – In this mode, a connection is made to a centralized PBX and then can ring down to one of many remote users.

The lifecycle of a VoIP Call Across a VoIP Network Between SIP Gateways

Successful gateway-to-gateway call setup and disconnect involve several protocols and concepts.  There are many steps that occur from the time when the calling party calls, the receiving party answers, and the call is terminated (refer to figure 5).  The type of phone equipment where the call originates from determines the path or network to be used for the initiation and completion of the call.  The Session Initiation Protocol (SIP) is an application-layer control protocol that creates, modifies, and ends sessions between users.  It is used in IP telephony and multimedia conferences. 

Description of Signaling System 7

Signaling System 7 (SS7) is an architecture that supports call-establishment, information exchange functions, routing, and billing mechanisms (illuminet®, 2004).  It utilizes out-band-signaling techniques within the public switched telephone network (PSTN).  Out-of-band signaling uses a band other the one used by the conversation.  While SS7 is used for signaling between network elements, there is also signaling between the subscriber and the switch that is accomplished using the ISDN D channel (IEC, n.d.).  The SS7 architecture (North America) is comprised of three essential components that are connected using signaling links (refer to figure 6).  The three components are:

  •  Signal Switching Points (SSPs) – These are telephone switches with SS7-capable software and are connected using signaling links.  They create, switch, and terminate calls.
  • Signal Transfer Points (STPs)– These are packet switches on the network.  They route signaling messages and perform other specialized routing functions.
  • Signal Control Points (SCPs)– These points are databases that provide necessary information for more advanced call-processing functions.

The SS7 protocol stack has 4 layers (levels) as defined in the OSI model.  The first three layers make up the MTP, and the fourth layer is the user part, the transport layer in the OSI model.  The ISDN User Part (ISUP) defines a protocol used in PSTN and uses the MTP for routing messages from one SSP to another.

Softswitch concepts and applications

The components in a VoIP network include central office switches, the SS7 network, media gateways, and a Softswitch configuration.  These components form both physical and logical control and data planes.  Logically, Softswitch concepts allow emulation of circuit switching in software.

 The term for an open API software package that is used to connect VoIP and PSTN networks is a Softswitch.  It consists of media gateway translation software and is a signaling gateway and media gateway control mechanism.  It provides an open layered architecture that controls services, applications, features, and transport hardware.  Soft-switched applications are less expensive than circuit-switched solutions, and allow room for innovation, while giving the user a choice of best-in-class hardware products.  The Softswitch can be broken into two distinct entities; the media gateway and the call agent.  A call agent receives signals from, and controls, multiple media gateways, ensuring an efficient and reliable converged network (see figure 9).

The functions of media gateways

Different physical networks are connected by the media gateway to provide end-to-end connectivity.  The media gateway can create a link between dissimilar devices and endpoints as a typical network switch can.  Media gateways are also used to connect PTSN circuits to VoIP networks, eliminating gaps, and allowing connectivity between different types of end devices; remaining transparent to users.

Voice over IP Trends and Advancements

Voice over IP systems have been used by the business sector for more than twenty years, and because of the powerful solutions offered and added flexibility systems will continue to be the choice businesses, both small and large (PCS VoIP, 2019).  The technology is constantly improving and will result in new trends for telecommunications.  Here are a few new trends that can be expected:

  • The quality of VoIP calls will improve with the deployment of 5G mobile networks.  With wireless speeds more than ten times faster than 4G, 5G will reduce common problems like echoes, packet loss, and call vibration.  Industries will benefit from the ability to utilize a remote workforce at greater volume.
  • Artificial intelligence (AI) will play an important role in systems development and will be used to analyze call quality before a call is placed and fix any quality issues immediately.  High bandwidth, correct network configuration, and low latency can be controlled with the need for minimal human interaction.
  • Services like VoIP and real-time video will be integrated into the cloud, mobile applications, and desktops.  In future years, if an application does offer a seamless voice communication service it will be considered archaic. 
  • Traditional landlines will become a thing of the past.  When AT&T is able to drop all old analog lines, old systems will die.  A legislature is currently in the works in Springfield, Illinois to allow AT&T to do just that.  Over 750,000 businesses in Illinois will be impacted if the FCC approves the legislation.

There is a larger technology that voice over IP is part of, Unified Communications (UC).  More specifically, the trend will be UC as a service (UCaaS) and will offer businesses improved collaboration and communication features. 

New hacking trends are expected this year, including an increase in eavesdropping, toll fraud, and Distributed Denial of Service (DDoS) attacks.  Because of SIP trunk vulnerabilities that can be exploited companies should use dedicated connections and ensure employees are using strong passwords.  Some companies are incorporating voice authentication techniques to avoid the issue of long complex passwords.  With VoIP solutions, this creates an additional vulnerability because once a person’s voice is recorded it can be used to create a computer model capable of generating any choice of words in the user’s voice.  The generated voice could be used for authenticated or in spoofing attacks involving finances and privileged information.

Conclusion

With the advancements in voice over IP technologies, the traditional telephone services and infrastructures are becoming unnecessary in many parts of the world.  As these services continue to disappear support for these legacy systems will remain in place for quite some time.  Standardized protocols are allowing wide-spread adoption of new voice communication technologies, both within wired and wireless networks.  Software solutions are replacing traditional circuit-switching methods and processes.  Industries prefer an open layer communication architecture to provide services and allow for the development of new applications, features, and hardware solutions.  Softswitch offers media gateway translation software and is an efficient signaling gateway and media gateway control mechanism.

During this phase of VoIP adoption, interconnectivity services and advanced layer 3 devices allow for seamless integration of the new technologies being created.  Future trends will improve the quality and efficiency of data communications on converged networks.  Unified Communication as a Service will allow SMBs and enterprises to experience the benefits of increased flexibility, lower maintenance, better compliance, and increased security.  The deployment of 5G technologies will offer higher bandwidth availability, and reduce common problems like excessive packet loss, voice echo, and call vibration. 

Artificial intelligence will play an important role in analyzing transmission quality and ensuring proper network configuration resulting in fewer man-hours expended and a better overall call experience for end users.  It will also assist with compliance and threat detection by monitoring communications and recognizing keywords related to privacy concerns and malicious behaviors.   

Technology gives us power, but it does not and cannot

tell us how to use that power. Thanks to technology,

we can instantly communicate across the world,

but it still doesn’t help us know what to say.

Johnathon Sacks

Updated 02/15/2021 by Kirby Allen