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eBook: Understanding SIP Trunking

Telling the whole story of SIP Trunking from the beginnings of the internet to managing modern SIP networks.

Chapter One: The Origins of SIP

The Internet as we know it began as ARPANET (a.k.a. the Advanced Research Projects Agency Network) in 1969. ARPANET started out as a four-node packet switching network through a government-awarded contract. It was initially deployed to support three main applications: e-mail, remote terminal access to host computers, and file transfers. The common thread running through these three applications is the characteristic of non-real-time operation. That means if there is a network delay of a few seconds, none of these functions will suffer any noticeable interruption of service.

SIP Trunking InternetAs part of the ARPANET research, the Internet Protocol (IP; providing addressing and packet delivery) and the Transmission Control Protocol (TCP; for end-to-end reliability between communication hosts) emerged in 1981. The creation of these protocols helped to transition the Internet to the state we know it in today – a worldwide collection of interconnected networks, supporting over a billion end-users across the globe.

Today, the Internet has grown to include real-time applications like voice and video. As a result, the standards organization that handles Internet process and protocol development – the Internet Engineering Task Force (IETF) – is focused on supporting these types of traffic. Even so, the basic infrastructure of the Internet – based upon the Internet Protocol (IP) standard and its basic structure as a connectionless, data-centric network – has not changed.

In today’s Internet full of real-time applications, running over a connectionless environment can cause interruptions in service like packet delay and loss, and a degraded experience for the end user. To fix these potential issues and prevent them from occurring, the IETF created a number of additional protocols like:

  1. Real Time Protocol (RTP)
  2. RTP Control Protocol (RTCP)
  3. Session Initiation Protocol (SIP)

SIP TransformsSIP Transforms

For the past 10 years, SIP in particular has played a key role in transforming the data-centric Internet into a broader infrastructure capable of handling both data-focused and real-time applications. This additional protocol has become a favorite of software developers, designers, and manufacturers.

To assist the IETF with the development and implementation of Internet technologies, a non-profit organization, the Internet Society, was founded in 1992. The Internet Society helps provide overall leadership to the international Internet community. Together, these two organizations help to guide the progression of Internet standards and improvements.

In delving deeper into the requirements for real-time protocols like SIP, we need to examine various options for network architecture.

There are two main communications network services options for end-user applications, connection-oriented and connectionless networks:

1. Connection-oriented Network

  • A classic example is the Public Switched Telephone Network (PSTN) – where the path from the source to the destination is established prior to any information transfer.
  • The signaling process uses resources but the path through the network is defined.
  • Transmission Control Protocol (TCP)

2. Connectionless Networks

  • A typical example is the U.S. Postal System – where a person drops a letter into the mailbox and if everything works as it should, the letter arrives to its intended destination.
  • Networking example is Ethernet Local Area Network (LAN).
  • Full source and destination address is attached to a packet of information.
  • Sometimes called “Best efforts” and unreliable.
  • Internet Protocol (IP) and User Datagram Protocol (UDP).

Both connection-oriented and connectionless networks require addressing to uniquely identify both the source and destination end stations. The addresses are made up of 48-bit Media Access Control (MAC) address IDs for their Ethernet hardware interface.

Although there are limitations that come with the connectionless nature of the Internet, they can be overcome through the use of additional protocols. Back in the 1970s, computer manufacturers and telecommunications companies designed proprietary architectures to try and maintain and grow their market share. But once the Open Systems Interconnection (OSI) Reference was published in 1984 by the International Organization for Standardization (ISO), this began to change. The OSI model broke the complex communication processes down into 7 layers that have been used by these manufacturers widely in the last few decades for standardization.

Similarly, the ARPANET developers divided communication processes down into four layers:

  1. Network Interface – physical connection to the LAN/WAN
  2. Internet – end-to-end delivery of datagrams or packets (IP)
  3. Host-to-Host – assures reliability of end-to-end datagram flow (TCP and UDP)
  4. Process/Application – providing end-user services like email or file transfer

This structure worked at the time that TCP/IP was developed, since applications like e-mail and file transfers were not real-time. In the 1970s there were no active server pages or YouTube videos or other time-sensitive applications. Support for real-time applications was not part of original design requirements outlined by ARPANET.

So fast-forwarding to now, how do we support connection-oriented applications like voice and video over a connectionless environment like IP without redesigning network infrastructure? The solution is to enhance IP with additional algorithms and protocols that fill in gaps.

These include such items as:

  • Audio and video encoders: codec functions defined in mathematical algorithms that are implemented in software or firmware
  • Real-time Transport Protocol (RTP): end-to-end delivery services for applications requiring real-time support, such as payload identification, sequence numbering, and timestamps
  • RTP Control Protocol (RTCP): monitors the quality of the RTP connection and provides information about the participants in the communication session
  • Session Description Protocol (SDP): conveys standard representation of communication details to session participants and information about the sessions media streams
  • Session Announcement Protocol (SAP): transmits packets periodically to identify open sessions of interest to the end-user community
  • Session Initiation Protocol (SIP): application layer control protocol used to create, modify, and terminate communication sessions between participants

SIP, specifically, is implemented frequently, because it bridges across wired, wireless, and Internet communications, and therefore can be used in today’s converged networks. Thus, supplementary protocols like SIP are enabling data-centric traffic and real-time conversations like voice and video to co-exist harmoniously.

Chapter Two: The Rise of SIP

As SIP has evolved into a popular protocol for establishing multimedia communications sessions, it is intriguing to look at the journey through the building blocks of SIP and its evolution into a standard telecommunications protocol of choice.

The Motivation Behind SIP

Signaling is the process that is undertaken to initiate a tele-communications session and then terminate the session when the parties have completed their business. SIP can generate signaling processes involving several entities and processes including: end users (humans or machines), switching systems, billing systems, etc. But before looking at the signaling processes that are provided by SIP, we should become familiar with the two main groups that brought about the standards that influenced VoIP technologies and signaling protocols like SIP. These groups are:

1. International Telecommunications Union (ITU)

  • Founded in Geneva, Switzerland during the 1860s, they produced agreements that supported connections between telegraph facilities across different countries and expanded into radio, TV, satellite, digital telephony, and VoIP.
  • Focus is circuit-switched communications
  • ITU has three main sectors:
    • Radio-communication (ITU-R) – manages available wireless spectrum
    • Telecommunications Standardization (ITU-T) – develops international networking standards
    • Telecommunications Development (ITU-D) – makes modern telecommunications services available to people in developing countries
  • Proposed ITU-T standards may take years before they are formulated into documents, as representatives from different countries on varying committees must review them before they are finalized.

2. Internet Society (ISOC)

  • Formed in 1992 as the global clearinghouse for Internet-related technologies.
  • Focus is packet-switching and data transmission
  • ISOC has four main groups: the Internet Architecture Board (IAB), the Internet Research Task Force (IRTF), the Internet Engineering Steering Group (IESG) and the Internet Engineering Task Force (IETF). The IETF’s tasks include:
    • Responsibility for developing and publishing Internet Standards called Request for Comments (RFC) documents
    • Drafting documents from the groups that are circulated online and discussed 3x per year at IETF’s Plenary Meetings
    • Completing documents that have been extensively reviewed and approved. These documents are given a number and available online via the RFC Editor (i.e. RFC 3261 the Session Initiation Protocol).

The IETF’s work on SIP is concentrated in the Session Initiation Protocol Core (sipcore) Working Group. This group’s mantra is to ensure that existing Internet protocols and architectures can be integrated with other Internet applications.

Key Elements of SIP

The baseline standard for SIP, RFC 3261, defines five different facets of establishing and terminating the multimedia communications that this protocol supports:

  1. User location: choosing which end system will be used for communications
  2. User availability: determining the willingness of the called party to engage in communications
  3. User capabilities: deciding which media and media parameters will be used during the session
  4. Session setup: establishing session parameters at both called and calling parties (when “ringing” occurs)
  5. Session management: including transfer and termination of sessions, modifying session parameters, and invoking services

To implement these five requirements, SIP architecture defines two main devices: clients and servers. A client can be a network entity that sends SIP requests and receives SIP responses, like a proxy device or user agent client (a logical function that creates a request and uses the client functionality to send that request). A server can be a registrar that accepts register requests and puts the information into the location server or possibly a redirect server, which re-routes the client to another server to finalize the client’s request.

SIP Addressing

In order to initiate and maintain a communication session, an identifier or specific address needs to be generated. This identifier is known as a SIP Uniform Resource Identifier (SIP URI). Resources that might need to be identified during this type of communication process include a user of an online server, a mailbox on a messaging system, an organizational group, or a phone number for a PSTN destination, which has to be conveyed to a gateway service. This SIP URI is like an email address and contains two parts, a username and hostname, like sip:[email protected].

SIP Messages

To continue the SIP communication session after addressing has occurred, SIP messages must be exchanged between clients and servers. The message types include such responses as REGISTER, INVITE, ACK, CANCEL, and so on. The message types provide the clients with responses from the servers that let them know which action the server is taking in regards to the client’s request.

The SIP session could include voice, video, and/or data information, so a standard method of describing the session being initiated is necessary (using the INVITE message type). This function is defined using the Session Description Protocol (SDP). The response messages provide return responses to these messages in the form of status codes and reason phrases. They can indicate success, redirection for further action, or errors, including global failure of the request to be fulfilled.

SIP Call Example

West Unified Communications Services delivers SIP-based calls to its customers through a process called SIP Trunking. This is a service that delivers phone communications services to customers purchasing an IP PBX. It is more flexible and efficient than traditional PSTN trunking service (which is PRI-based), since it can reduce service costs, provide disaster recovery, and enable real-time call management reporting. This SIP Trunking service example uses SIP as a way to construct each call. Here are the basic steps:

  1. Remote party places call through PSTN
  2. West connects to the PSTN via SIP using media gateways that are serviced by several nationwide local and long-distance providers
  3. The connections interface with West’s MaxxConnect IP Trunking solution resulting in an SIP connection.

From its birth in the IETF, to its development and growth into a protocol of choice for multimedia communications sessions, SIP has grown into a standard for the present and future of telecommunications operations.

Chapter Three: Components of a SIP-based Network Architecture

WAN technologies have migrated from analog environments to digital environments in the past few decades. With this transition, the cloud is now positioned to better support SIP-based network infrastructures. Alongside the evolution of the WAN, three new classes of networking devices have been developed. These networking devices are:

  1. The Softswitch
  2. The IP-PBX
  3. The Session Border Controller

The functions and capabilities of these devices will be explored in light of their support for SIP-based networking environments.

The Impact of the Internet and SIP on WAN

In the mid-1990s, the Internet Engineering Task Force began its development of the Session Initiation Protocol. SIP was to the voice world what the Hypertext Transfer Protocol (HTTP) was to the World Wide Web – another application that could be run over an IP-based network – and one that would consume additional WAN bandwidth.

The New Architecture: SIP-Based Switching and SIP Trunks

With SIP becoming an industry standard, and with the exponential growth of bandwidth consumption within most organizations, a need arose for a technology solution that could meet all requirements. This solution was a SIP-based trunk that allowed remote locations to connect directly without all of the protocol conversions. The evolution of this technology is further proof that the world is moving in the direction of SIP.

Benefits of SIP TrunkingBenefits of SIP Trunking

There are many benefits to implementing SIP Trunking. It can save an enterprise money, since SIP trunks can be purchased in the exact increments needed. Furthermore, as offices may grow or downsize, SIP trunks can be scaled to any specific capacity – up or down. Another benefit is that the SIP trunk can be adjusted for changes in bandwidth requirements, such as for seasonal call volume increases. Moreover, SIP is a necessary platform for running Unified Communications applications like presence. Finally, SIP trunk lines provide an increased level of reliability and business continuity. For instance, SIP trunk lines can be programmed to reroute to backup sites if the need arises.

Softswitches: The New Standard in Switching

The Public Switched Telephone Network and Internet can be likened to a freeway – there are specific paths that each call takes from source to destination and a great deal of traffic along the way. On this journey, there are three basic functions that are required: routing, transmission, and billing. While the PSTN has understood and utilized these functions for years, they needed to be adapted for Internet VoIP technologies in order for them to be fully operational on an end-to-end basis. For the PSTN, these functions are primarily accomplished through the use of switches that are strategically placed around the world. The switches are made up of two key elements:

  1. Switching fabric – makes the physical connection
  2. Switching logic – provides call routing and control functions, custom calling features, and interfaces to other systems like billing

Switches have gone through an evolutionary process, starting out in the form of telephone operators, then relay-based mechanical switches, and later developing into fully-electronic switches. A new generation of switches, known as softswitches, are solving many of today’s network migration challenges, providing a path to the future of integrated voice, video, and data applications. Softswitches help in the following ways:

  • Replace most of the hardware switching with computer software-based functions
  • Separate the physical switching function from the control logic function, enabling devices in a variety of locations to perform these operations
  • Provide mechanisms to bridge the connectivity gap between existing circuit-switched (PSTN) and evolving packet-switched (VoIP) services

IP PBXSoftswitch Architecture

Softswitches are made up of two major elements: a call agent and a media gateway. The call agent provides most of the network intelligence, handling the call routing, network signaling, billing, and related functions.

The media gateway delivers the physical connection for an end-to-end path, incorporating a variety of LAN and WAN interfaces like Ethernet, IP, T1/T3, ATM, etc. The call agent controls the media gateways across the globe, and the type of transmission formats such as analog, LAN, WAN, and so on, that are involved.

Softswitch functions can be divided into four architectural areas:

  1. Transport – carries the call signaling and media (voice, data, video) signals
  2. Call Control and Signaling – controls the devices in the VoIP network and establishes and disconnects media sessions
  3. Service – controls services and applications like applications servers
  4. Management – deals with service provisioning, support, and billing

In order to connect the logical and physical sides of the softswitch architecture, communication must take place between the media gateway controller and media gateway. This function is accomplished when the media gateway controller sends commands to the media gateway using one of two protocols – either the Media Gateway Control Protocol (MGCP) or the Media Gateway Controller (MEGACO) protocol.


Another key networking device for SIP-based environments is the Private Branch Exchange (PBX). The PBX has been the workhorse of premises-based communications for several decades. It acts as a communication circuit between each end station or an end station line and outgoing trunk circuit. Furthermore, it calls for a limited number of trunk circuits to the central office that are shared amongst the end stations. The PBX facilitates this trunk sharing.

PBXs are installed at customer sites to provide switching for end-user stations. This equipment connects end-user stations to one another either onsite or to offsite stations on the company’s private network. They can also connect end-user stations to other destinations such as cell phones, which are accessible through the PSTN. PBX technology has gone through the following evolutionary stages: from mechanical switching systems, to analog systems, to digital switches supporting both data and voice, to the current stage of switching technology containing an infrastructure that supports IP technologies.

The PBX is made up of various end-user features. Smaller switching systems called Key Telephone Systems (KTS) provide basic features like caller ID, call hold, call transfer, voicemail, etc. These systems are designed for small organizations with 5 to 50 end stations. Larger PBX systems supporting tens of thousands of end stations provide additional features, including an auto-attendant, conference calling, call detail recording, call center capabilities like call queuing or statistics, and can integrate with other systems like Microsoft Outlook.

Session Border Controllers

A final SIP-based WAN network access product, the Session Border Controller (SBC), provides the following capabilities:

  • Network-to-network security
  • Resolves IP addressing differences between networks
  • Keeps track of key call information like calling and called parties and the call duration
  • Allows VoIP calls to traverse Network Address Translation (NAT) devices and security firewalls
  • Provides admission control
  • Monitors Quality of Service (QoS) and Service Level Agreement (SLA) issues
  • Provides protocol translation between different networking formats

SBCs function as a demarcation point between two networks – an inner network and an outer network. The architecture consists of two components: a signaling component and a media component, which both logically and physically connect the two networks. The signaling piece deals with service requests like connects, disconnects, authentication, and authorization. The media piece deals with the transport of information including QoS and SLA monitoring.

To keep up with the evolution of the WAN and corporate bandwidth demands, advanced networking technologies like softswitches, IP-PBXs, and Session Border Controllers have developed, providing more capabilities and better functionality for today’s changing networks. Moreover, it has been estimated that well over half of all companies will be implementing SIP trunking by the end of 2015. As a result of this increase in adoption comes a need to delve further into the many concepts surrounding SIP Trunking.

Chapter Four: Concepts of SIP Trunking

As interest in the Session Initiation Protocol has grown, new and innovative applications have been developed. One of these is the SIP trunk, which can be used to connect IP PBX systems with service provider networks, and more effectively replace traditional trunks that are based on Time Division Multiplexing (TDM) technologies.

The SIP Trunking Environment

For most enterprises, unless you have a 100% wireless network, some type of hardwired connection is required to the PSTN. If the network is comprised of TDM-based architecture, high-speed circuits like T1 circuits or ISDN PRI circuits running at 1.544 Mbps are used to connect switching systems like PBXs to the service provider. Theses circuits are called trunks because they provide a direct pipe from the enterprise to the outside world. The SIP trunk provides connectivity functions like channel capacity, call statistics, and so on, more effectively than previous technologies.

Legacy network architectures were based on TDM technology that had separate networks for voice (based on TDM) and data (IP). With the evolution of converged networks, TDM-PBXs have been replaced with IP-PBXs, moving all communications to an IP-based platform. However, in order to communicate outside enterprise walls using purely IP, trunks from a service provider are still necessary. And if the trunks are based on legacy TDM technology, a gateway is required between the company’s IP PBX and the carrier circuit to provide functions like data protocol and transmission rate conversions. With a system that includes multiple protocols and their resultant conversions, signal processing occurs – and along with it, delay and distortion issues in the end-to-end communications path.

SIP Trunking: Enabling Growth

Traditional T1 or ISDN PRI trunks limited network growth to twenty-four channels at a time – one for signaling and the other twenty-three for end user voice or data traffic. Once a customer used up this capacity, they were forced to order up another twenty-four channels even if only one or two were required, resulting in unnecessary costs for the company. But, since SIP trunks are provisioned separately, you only pay for what you need – five channels = five trunks…no more, no less. Furthermore, SIP Trunking provides further cost-savings since no gateway is required; the ITSP includes a SIP-compatible trunk, so no additional hardware is required. Finally, since the SIP trunk provides both data and voice connectivity, you get two networks for only one service.

SIP Trunking: The Need for Speed

Today’s need for real-time communications like voice and video don’t perform well with these kinds of delays. The fix: Internet Telephony Service Providers (ITSPs), carriers that have incorporated IP instead of legacy TDM technology as the backbone of their network architecture. ITSPs provide enterprises with an IP-based trunk that is compatible with SIP call signaling, thus eliminating the protocol conversions that can cause network latency.

SIP Trunking Benefits: The List Goes On...

SIP Trunking provides a number of benefits to the enterprise:

  • Converged data and voice networks = economies of scale and better Quality of Service (QoS) management
  • Location independent
  • No gateway required between the IP PBX and the WAN = less time required for installation and administration
  • Flexibility to reallocate software-based IP-PBX for bandwidth-intensive processing like video conferencing
  • ITSPs can focus on IP only instead of trying to meet the needs of varied network architectures

In order to provide all of these benefits and more, there is a need for implementation standards that are designed to reduce interoperability challenges within each network.

SIPConnect: A Standards-based Implementation

The SIP standards were originally developed by the IETF and published as RFC 3261. But in order to flesh out all of the details required for successful implementations, an industry group known as the SIP Forum – comprised of both equipment vendors and next generation carriers – was created. The SIP Forum’s mission is: to advance the adoption of products and services based on the Session Initiation Protocol. The group goes about their mission in these five ways:

  1. Organizing interoperability and SIP testing events
  2. Developing industry-wide best-practices implementation guides
  3. Creating documentation supporting the deployment of SIP-related technologies
  4. Building awareness of the benefits of SIP through educational seminars
  5. Acting as a central information clearinghouse for the SIP-related industry

So, while the IETF sets the standards, once the standard is complete, they pass the baton to the SIP Forum to clarify and work towards the resolution of any outstanding technical issues. In this way they help to provide independent vendors interoperability with their systems.

The SIPconnect 1.1 Technical Recommendation is a profile of the Session Initiation Protocol (SIP) that enables direct connectivity between a SIP-enabled Service Provider Network and a SIP-enabled Enterprise Network. It specifies the set of IETF and International Telecommunications Union – Telecommunications Sector (ITU-T) standards that must be supported, provides precise guidance in the areas where the standards leave multiple implementation options, and specifies a minimal set of capabilities that should be supported by the Service Provider and Enterprise Networks. To increase compliance and interoperability with SIP, SIPconnect certifications are available for eligible vendors.

SIPConnect Architecture

The architecture required for both service provider and enterprise network interoperability using SIP includes the following:

  • SIP servers and gateways to the PSTN (enterprise end) and Signaling System 7 (SS7) networks (service provider end)
  • SIP servers, IP PBX, and IP phones are also required on the enterprise side
  • Four protocols and systems are involved:
    • SIP
    • Real Time Protocol (RTP) – carrying voice samples
    • TDM – providing framing format required by the PSTN
    • SS7 – enabling call setup on the PSTN

Other specific requirements that may apply to the enterprise, service provider, or both include: locating SIP servers, signaling security, firewall and Network Address Translation (NAT) transversal, authentication and accounting, enterprise PSTN identities, URI formatting and addressing rules, Quality of Service, media attributes, and PSTN Interactions – all of which are specifically outlined in the SIPconnect technical recommendation.

Chapter Five: Managing the SIP Network

As deployment of SIP-based systems has increased, so has the need to manage these complex systems. As such, we will be reviewing the different facets of network management and how they apply to the challenge of managing a multimedia network.

Network Management for Multimedia Networks

Network performance is often an afterthought for most companies until something goes wrong. When a network failure occurs, network management and performance then becomes a higher priority. So, to assist companies with network management, the International Standards Organization developed a management framework in 1989 to integrate with the Open Systems Interconnection Reference Model. This model divides network management tasks into five specific management functions and defines each accordingly:

  1. Fault Management – encompasses fault detection, isolation, and correction of abnormal OSI environment operation.
  2. Accounting Management – enables charges to be established for the use of resources in the OSI environment, as well as user resource costs identified.
  3. Configuration Management – identifies, exercises control over, collects data from, and provides data to open systems for the purposes of preparing for, initializing, starting, providing for the continuous operation of, and terminating interconnection services.
  4. Performance Management – enables the behavior of resources in the OSI environment and the effectiveness of communication activities to be evaluated.
  5. Security Management – supports the application of security policies via functions which include the creation, deletion, control of security services and mechanisms, and the distribution of security-relevant information and reporting of security-relevant events.

To optimize the network in general, and specifically for multimedia devices, enterprise managers should evaluate the status of their network based on the five functional areas above and determine whether modifications should be made.

Quality of Service (QoS)

An important part of network management is ensuring end users are satisfied with the way the network is operating, and the overall quality of service it provides. Reviewing where the concept of network quality came from, and what modifications may need to be made in regards to quality of service, is especially crucial when dealing with more complex multimedia environments and the needs unique to real-time applications.

The original benchmark for quality in the telecommunications industry began with AT&T. When AT&T monopolized the telecommunications market, one of its main performance standards was known as the Five Nines of Reliability. At that time, the quality objective for central office switching systems specified two hours of downtime in 40 years of operation. Working through all the time conversions, you arrive at what we know today as 99.999% reliability (the five 9s) factor.

Defining Quality of Service (QoS)

The ITU-T defines the subjective term Quality of Service as “the collective effect of service performance which determines the degree of satisfaction of a user of the service.” The operative words here are end-user and satisfied. In other words, the end users’ performance criteria for the multimedia network must be met.
As a result, much research has gone into the subject of QoS resulting in a number of algorithms and protocols that have been developed to address and meet user satisfaction for quality issues.

Meaduring Toll Quality

So what is toll quality for voice systems and how is it measured? The ITU-T has addressed these questions in two key recommendations:

1. Methods of Subjective Determination of Transmission Quality (P.800)

  • Consists of the Conversation Opinion Test – volunteers are asked to render an opinion of the connection used based on a 5-point scale, ranging from 1 (bad) – 5 (excellent).
  • A large number of test subjects are used with an average taken of the individual scores = the Mean Opinion Score (MOS).
  • A MOS of 4 is considered a target toll quality within the telephone industry.

2. Perceptual Evaluation of Speech Quality (PESQ), an Objective Method for End-to-end Speech Quality Assessment or Narrow-band Telephone Networks
and Speech Codecs (P.862)

  • Addresses the effects of filters, variable delay, and coding distortions
  • Used for both speech codec evaluation and end-to-end measurements
  • Implemented in hardware for greater accuracy

Using both QoS testing and evaluation methods can provide a bird’s eye view of the overall voice system quality.

Quality of ServiceOptimizing QoS on the WAN Using MPLS

Multiprotocol Label Switching (MPLS) originated from Asynchronous Transfer Mode (ATM) in the mid-1990s, when the Internet Engineering Task Force and others sought to integrate the high speed Layer 2 switching in ATM with the Internet Protocol (IP) Layer 3 routing technologies. A key reason for the interest in MPLS is how easily it can be applied and implemented over a wide variety of networking infrastructures, including ATM, Frame Relay, LANs, etc. For instance, MPLS-capable routers called Label Switching Routers (LSRs), only have to examine the MPLS header and not the entire IP header, so the QoS mechanism can operate independently of the Network Layer protocol.

Impairments that Impact Voice Quality

In terms of factors that influence voice quality, it is important to remember that PSTN is circuit switched and connection-oriented, while VoIP networks are packet switched and connectionless. With the differences in infrastructure, varying QoS challenges can occur. For example, several of the VoIP network QoS challenges come from using packet switched networks to perform functions that they weren’t originally designed to support, like voice transmission. Because of the packet conversion that takes place between the technologies, there can be transmission impairments to the end-to-end network voice connection. Other items affecting voice QoS:

  1. Packet Delay or Latency – the difference between when the signal is transmitted and when it is received
  2. Packet Jitter – the variation in arrival rates between successive packets
  3. Packet Loss – the measure of the number of packets from the original data stream that don’t find their way to the destination

Of these three main issues with voice QoS, packet delay is most often the focus. In fact, the ITU-T published a benchmark for voice circuit delay that details accepted delay objectives. In their recommendation, delays of 150-250 milliseconds per one-way transmission path are within acceptable standards. Charts contained with the ITU-T’s G.114 standard are available to assist VoIP engineers with necessary design variable modifications, along with design tools that can assist with delay budget calculations.

Tools for VoIP Network Management

There are many challenges that correspond with converged networks. Fortunately, there are several management tools that can be used to
assist in identifying and solving these issues.

SNMP Network Management Systems

  • Run on a protocol suite developed by the IETF that first became popular in the 1990s
  • Two key elements
    • Manager (console) – oversees the network operations over UNIX or a similar platform
    • Agent – monitors the local operations and reports key data back to the manager

Protocol Analyzers

  • Used by network administrators and engineers
  • Attaches to the LAN/WAN and captures all data and voice packets travelling along the wire, analyzes the packets based on the protocol being used, then displays that information in various formats Performance/Quality Monitors
  • Look at network operations independently from underlying protocols in use
  • Display high-level information providing a snapshot of network health containing statistics like Mean Opinion Score, packet delay, etc.

Optimizing SIP Trunking Network Solutions with West Unified Communications Services

West has been delivering Unified Communications (UC) services to clients over a next-generation MPLS network since 2000. Its managed MPLS networking service is flexible, scalable, and reliable; it allows enterprises to replace separate data, video, and voice networks with one affordable, converged network.

For companies looking to reduce the costs of telephony through the deployment of SIP trunking services, West offers a fully managed solution with broad geographic coverage, and support for a wide range of network carriers and PBX systems. MaxxConnect enables a single solution backed with industry-leading business continuity tools, advanced call routing capabilities, and expert support that ensures a company reaps the full benefits of IP trunking.

The MaxxConnect suite of IP trunking solutions is designed to provide enterprise clients with the carrier–grade service they have come to expect from traditional providers, along with the benefits of next–generation IP–based service that allows their business to run more efficiently. It delivers a consistent set of voice services across the enterprise infrastructure, with flexible IP and TDM Trunking options for the company’s on-site PBX. MaxxConnect is delivered over virtually any MPLS network, like West’s own Maxxis network, providing the efficiency and economic benefits of network convergence and eliminating the need for local PSTN gateways and costly PRIs.

Moreover, West utilizes industry best practices for optimum interoperability with a variety of IP-PBX systems, and maintains certified partner status with both Cisco, ShoreTel, and Avaya to deploy SIP Trunking to their respective IP-PBX platforms. In addition, these customized trunking solutions – integrated with components from industry leaders like these – come complete with sophisticated Disaster Recovery and Survivability features.

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