An Extensive Telecom & CCaaS Glossary

An Automatic Call Distributor is a telephony system that intelligently routes incoming calls to the most appropriate agent or department based on predefined criteria such as agent skill, availability, language, or customer history. In modern cloud contact centres, the ACD operates as the routing brain of the operation — ensuring customers reach the right person as quickly as possible.

In a Genesys Cloud CX environment, the ACD works alongside AI and predictive routing engines to dynamically match each incoming interaction to the best-fit agent. This goes beyond simple round-robin distribution; it factors in real-time queue conditions, customer sentiment signals, and historical interaction data. The result is shorter handle times, better first-contact resolution, and measurably higher customer satisfaction scores. Star Telecom’s SIP trunking infrastructure is purpose-built to deliver calls to your ACD with the audio quality and low latency that modern routing engines require.

Average Handle Time is the average duration of a single customer interaction from the moment an agent answers to the moment the call or interaction is fully wrapped up, including after-call work. AHT is one of the most tracked contact centre KPIs because it directly affects staffing costs and customer experience.

A high AHT may indicate agent knowledge gaps, inefficient CRM access, or unclear call flows. A low AHT is not always positive — rushed interactions reduce customer satisfaction. The goal is an optimized AHT where customers feel heard and issues are fully resolved. AI-assisted tools in Genesys Cloud CX, such as Agent Copilot, can reduce AHT by surfacing the right information to agents in real time, without sacrificing interaction quality.

An analog phone is a traditional telephone that transmits voice as a continuous electrical signal over a copper wire. While largely replaced by VoIP and digital systems in enterprise environments, analog phones remain common in legacy installations and certain emergency line configurations. When migrating from analog to SIP trunking, businesses typically use an FXS (Foreign Exchange Station) adapter to connect existing analog devices to a VoIP network, preserving hardware investment while gaining the cost and flexibility benefits of cloud telephony.

An Auto Attendant is an automated telephone menu system that answers incoming calls and routes them to the appropriate department, extension, or voicemail without requiring a live receptionist. Callers navigate the menu by pressing keypad digits (DTMF) or, in more advanced systems, speaking their request aloud.

Modern auto attendants in platforms like Genesys Cloud CX go well beyond basic menu trees. They integrate with CRM data to recognize returning callers, offer personalized routing options, and can hand off seamlessly to a live agent with full context. A well-designed auto attendant reduces call misrouting, lowers hold times, and creates a professional first impression. The underlying call delivery from Star Telecom’s SIP trunks must be low-latency and high-fidelity for DTMF detection and speech recognition to function accurately.

Bandwidth is the maximum rate at which data can be transmitted over a network connection, typically measured in megabits per second (Mbps) or gigabits per second (Gbps). For VoIP and SIP trunking, adequate bandwidth is essential to call quality — each active voice call requires approximately 64–100 Kbps depending on the codec used.

Before deploying SIP trunking, organizations should conduct a network readiness assessment to confirm available bandwidth supports the expected number of concurrent calls. A business with 50 simultaneous calls needs roughly 5–10 Mbps dedicated to voice traffic. Insufficient bandwidth causes degraded audio, call drops, and jitter. Star Telecom assists clients in right-sizing their trunk capacity relative to their actual Genesys Cloud CX call volumes.

Bring Your Own Carrier (BYOC) is a deployment model that allows organizations to connect their own preferred SIP trunking provider directly to a cloud contact centre platform, rather than using the platform’s bundled carrier. Genesys Cloud CX supports BYOC natively, giving enterprises full control over their telecom costs, carrier relationships, and routing architecture.

BYOC is particularly valuable for large enterprises that have negotiated favourable carrier terms, need specific geographic coverage, or require compliance with local telecommunications regulations. Star Telecom is a certified BYOC SIP provider for Genesys Cloud CX. Our direct peering architecture ensures audio arrives at the Genesys media layer with the quality and latency characteristics that AI-powered contact centre features — including real-time transcription and sentiment analysis — require to function correctly. BYOC with Star Telecom eliminates the carrier markup built into bundled carrier pricing, typically reducing per-minute and trunk costs by 30–40%.

A call flow is the defined path a call or interaction takes from the moment it enters the system to its final resolution — whether that’s connection to an agent, self-service completion, voicemail, or another outcome. A well-designed call flow minimizes transfers, reduces hold time, and ensures every caller reaches the appropriate outcome as efficiently as possible.

In Genesys Cloud CX, call flows are built visually in the Architect tool, allowing contact centre designers to create complex routing logic using drag-and-drop components. Star Telecom works with clients to ensure the SIP trunking layer is correctly configured to honour the call flow’s DID and toll-free routing requirements, including disaster recovery scenarios where backup routing paths must activate automatically.

Contact Centre as a Service (CCaaS) is a cloud-based software model that delivers all the capabilities of an enterprise contact centre — including ACD routing, IVR, workforce management, analytics, and AI — through a subscription-based, hosted platform. Unlike on-premises PBX systems, CCaaS requires no hardware to purchase, maintain, or upgrade.

Genesys Cloud CX is the leading CCaaS platform for enterprise and mid-market contact centres. It provides omnichannel routing across voice, chat, email, and social, combined with native AI tools for agent assistance and self-service automation. Star Telecom is a specialized SIP trunking partner for Genesys Cloud CX, providing the certified BYOC connectivity that ensures enterprise-grade voice quality within the CCaaS environment. The total cost model for CCaaS deployments depends heavily on the telecom layer — choosing the right SIP provider is one of the highest-impact cost decisions in a Genesys deployment.

A Call Detail Record is a data record generated by a telephone exchange or a SIP trunking platform that documents the details of every telephone call — including originating and terminating numbers, call duration, start and end time, and call outcome. CDRs are used for billing reconciliation, compliance auditing, traffic analysis, and fraud detection.

In a Star Telecom deployment, CDRs are accessible via the customer portal and can be exported for integration with billing systems or Genesys Cloud CX reporting dashboards. Accurate CDR data is essential for validating carrier invoices and identifying unusual traffic patterns.

A Competitive Local Exchange Carrier is a telephone company that competes with the established, dominant local telephone carrier (the ILEC) in a given market. CLECs typically offer services including local telephone service, long distance, and broadband using a combination of their own infrastructure and lines leased from the incumbent carrier.

Star Telecom operates as a CLEC in Canada, which means we hold our own carrier licenses and direct interconnections with the PSTN. This is the foundation of our ability to offer certified SIP trunking for Genesys Cloud CX without the middleman fees that resellers charge. CLEC status means our clients receive direct carrier pricing and regulatory protections.

CNAM (Caller ID with Name) is the service that displays a caller’s name alongside their phone number on the recipient’s telephone or screen. CNAM data is stored in a database queried in real time when a call is connected. For businesses, having an accurate and recognizable CNAM entry is important for call answer rates — calls from unknown numbers are increasingly screened or declined.

Star Telecom manages CNAM registration for client DIDs as part of our SIP trunking service, ensuring outbound calls display a recognizable business name and improving contact rates for outbound contact centre operations.

Computer Telephony Integration (CTI) refers to the technology that enables computers and telephone systems to interact in coordinated ways. In a contact centre context, CTI powers screen pops — the automatic display of a customer’s record in the CRM or helpdesk system at the moment a call is connected to an agent, based on the incoming caller ID or IVR-collected data.

CTI eliminates the need for agents to manually search for customer records, reducing handle time and enabling more personalized service. Genesys Cloud CX provides deep CTI integration with Salesforce, ServiceNow, Microsoft Dynamics, and other major CRM platforms through native connectors. Star Telecom’s SIP trunking passes caller ID data accurately through the PSTN interface to ensure CTI lookup logic fires reliably on every call.

Direct Inward Dialing (DID) refers to telephone numbers assigned to an individual, department, or contact centre queue that can be called directly without going through an operator or auto attendant. In a SIP trunking deployment, DIDs are virtual numbers provisioned by the carrier and mapped to specific extensions or call flows in the PBX or CCaaS platform.

Star Telecom provisions DID numbers across Canada and the US, including toll-free numbers with geographic and non-geographic area codes. DIDs are the foundation of call routing architecture — every IVR entry point, department queue, and agent extension is typically built on an assigned DID. During a porting project, existing DIDs are migrated from a previous carrier to Star Telecom with no change to the numbers themselves, ensuring business continuity.

DTMF is the signaling system used when you press a button on a telephone keypad. Each key generates two simultaneous audio tones at specific frequencies — their combination identifies which digit was pressed. DTMF tones are the mechanism by which callers navigate IVR menus (“Press 1 for Sales, Press 2 for Support”).

In VoIP and SIP trunking deployments, DTMF signals can be carried in three ways: in-band (as audio), out-of-band via RFC 2833, or via SIP INFO messages. Incorrect DTMF configuration is one of the most common causes of IVR breakdowns — digits fail to register, or the wrong options are triggered. Star Telecom uses RFC 2833 by default, which provides the most reliable DTMF delivery in IP network environments, ensuring Genesys Cloud CX IVR menus function correctly on every call.

E.164 is the ITU-T international numbering standard that defines the format for telephone numbers used in the global PSTN. An E.164 number is formatted as a plus sign followed by the country code, then the subscriber number, with no spaces or dashes (e.g., +16478550001 for a Canadian number). Most SIP platforms, including Genesys Cloud CX, require phone numbers to be provided in E.164 format for routing to function correctly.

When configuring DID numbers and outbound caller ID in a Star Telecom SIP trunk, all numbers must be in E.164 format. Incorrect number formatting is a common source of call routing failures during initial deployment.

Enhanced 911 (E911) is the emergency calling service that automatically provides a caller’s location information to emergency dispatchers when 911 is dialed. For traditional landlines, location is determined by the physical address registered to the phone line. For VoIP and cloud phone systems, location must be explicitly registered and kept current, since calls can be made from any internet connection.

In Canada, the CRTC mandates that all VoIP service providers deliver a dispatchable location with every 911 call. Failure to comply exposes organizations to regulatory risk and, more critically, delays emergency response. Star Telecom partners with RedSky to provide certified E911 services for cloud and SIP deployments, including Genesys Cloud CX. Every DID and extension in a Star Telecom deployment can be associated with a precise civic address, and the location database updates automatically as organizations move or expand.

In telecommunications, encryption refers to the process of encoding voice and data transmissions so they cannot be intercepted or read by unauthorized parties. For SIP trunking, two encryption standards are most relevant: TLS (Transport Layer Security) for protecting SIP signaling, and SRTP (Secure Real-time Transport Protocol) for encrypting the actual voice media stream.

Star Telecom supports TLS/SRTP on all enterprise SIP trunks. Organizations in regulated industries — healthcare, financial services, government — should ensure both signaling and media encryption are enabled. Genesys Cloud CX supports encrypted SIP trunking natively, and Star Telecom’s trunk configuration guides cover the exact parameters required for a fully encrypted deployment.

Failover in telecommunications is the automatic process of rerouting voice traffic to a backup carrier path, data centre, or routing destination when the primary path experiences a fault, outage, or degraded performance. True failover is instantaneous — callers should not experience any perceptible interruption in service.

For contact centres, failover is a business-critical capability, not a premium feature. A contact centre that is unreachable during a carrier outage loses revenue, erodes customer trust, and — in regulated industries — may face compliance penalties. Star Telecom’s SIP trunking platform includes automated failover as a standard component, not an add-on. Our multi-carrier architecture monitors path health in real time; if primary carrier latency exceeds threshold or packet loss is detected, traffic is rerouted to a redundant path within milliseconds. Toll-free and DID numbers are both covered. Failover routing can be configured to redirect to a backup contact centre site, a mobile number, or a recorded message, depending on the failure scenario.

First Contact Resolution (FCR) is the percentage of customer issues resolved in a single interaction, without the customer needing to call back, be transferred, or follow up through another channel. FCR is widely regarded as the most direct indicator of contact centre effectiveness — it measures whether the contact centre is actually solving problems, not just handling volume.

Improving FCR requires well-trained agents, effective knowledge management, accurate routing (so customers reach agents equipped to help), and reliable call quality. AI tools in Genesys Cloud CX — including real-time agent guidance and next-best-action recommendations — directly improve FCR by surfacing the right information at the right moment. Audio quality from the SIP layer also affects FCR: poor call quality forces customers to repeat themselves and creates miscommunication that extends interactions.

G.711 is the ITU-T standard audio codec used for uncompressed voice encoding in telephony. It encodes audio at 64 Kbps and produces the highest-fidelity voice quality of the standard telephony codecs. G.711 is the preferred codec for Genesys Cloud CX AI features including real-time transcription, sentiment analysis, and voicebot interactions, because compressed codecs introduce audio artifacts that reduce transcription accuracy.

Star Telecom uses G.711 as the default codec for enterprise SIP trunks. For AI contact centre deployments, maintaining G.711 throughout the media path — from the PSTN through the SIP trunk to the Genesys media server — is essential. Using compressed codecs like G.729 degrades transcription accuracy and increases AI token error rates, raising the cost of AI-assisted interactions.

G.729 is a compressed voice codec that encodes audio at 8 Kbps, compared to G.711’s 64 Kbps. While G.729 conserves bandwidth, it achieves compression through lossy audio processing that introduces artifacts and reduces voice fidelity. For traditional voice calls, G.729 is acceptable. For AI-powered contact centre applications that rely on accurate transcription and natural language processing, G.729 is problematic — the reduced audio quality meaningfully degrades recognition accuracy and increases AI processing errors.

Star Telecom recommends G.711 for all Genesys Cloud CX deployments with AI features enabled.

Geographic redundancy (geo-redundancy) means hosting critical infrastructure across multiple physically separate data centres in different geographic locations. If one data centre is affected by a power outage, natural disaster, or network event, the redundant site maintains service without interruption.

Star Telecom’s SIP trunking platform is geo-redundant across multiple Canadian data centres. This means your SIP trunk registrations, routing tables, and call processing logic are replicated in real time. Combined with automated failover, geo-redundancy provides the foundation for true 99.999% uptime — the standard required for enterprise contact centre operations. Genesys Cloud CX’s own cloud infrastructure is also geo-redundant; pairing it with a geo-redundant SIP provider closes the final single point of failure in the architecture.

H.323 is an older ITU-T standard for multimedia communications over IP networks, covering voice, video, and data conferencing. H.323 was the dominant VoIP protocol before SIP became the industry standard. Most modern contact centre platforms, including Genesys Cloud CX, use SIP exclusively. H.323 systems may still be found in legacy enterprise telephony environments. Migration from H.323 to SIP-based infrastructure is a common project scope for Star Telecom’s professional services team.

HD Voice (also known as wideband audio) refers to voice calls encoded at a higher frequency range than traditional narrowband telephony. Standard telephony transmits frequencies between 300–3,400 Hz; HD Voice extends this to 50–7,000 Hz or beyond, producing noticeably clearer, richer audio with less fatigue over long calls.

In contact centre environments, HD Voice improves agent and customer experience and reduces the cognitive load of listening in noisy environments. More importantly for AI applications, HD Voice codecs such as G.722 provide better quality audio for transcription engines, reducing word error rates. Star Telecom supports HD Voice on all modern SIP trunks.

High Availability (HA) refers to a system design that ensures a service remains operational and accessible for a very high percentage of time — typically expressed as a percentage such as 99.99% or 99.999% (often called “five nines”). For a contact centre, HA means callers can always reach your business, agents can always connect to the platform, and no single component failure causes a complete outage.

Achieving high availability in a SIP trunking context requires redundant carrier paths, geo-redundant infrastructure, automated health monitoring, and tested failover procedures. Star Telecom’s HA architecture provides automatic trunk failover, redundant PSTN interconnects, and real-time monitoring with 24/7 NOC support. Genesys Cloud CX is architected for HA natively — paired with Star Telecom’s HA SIP trunking, the combined solution delivers the uptime characteristics required for mission-critical contact centre operations.

The Incumbent Local Exchange Carrier is the established, dominant telephone company in a given local service area — the carrier that originally held the monopoly on local telephone service. In Canada, the major ILECs are Bell, TELUS, and SaskTel. ILECs own and operate the core telephone network infrastructure, including local loops, central offices, and PSTN interconnections. CLECs like Star Telecom lease portions of this infrastructure and compete directly on service and price, offering businesses an alternative to paying incumbent carrier rates.

Interoperability is the ability of different telecommunications systems, platforms, and devices to communicate and work together seamlessly. In SIP trunking, interoperability means the SIP trunk provider’s infrastructure is tested, certified, and proven to work correctly with the customer’s CCaaS or PBX platform.

Not all SIP providers are created equal in this regard. A SIP trunk that is certified for Genesys Cloud CX has been specifically tested against Genesys’s media layer, codec requirements, DTMF handling, and failover behaviour. Star Telecom holds Genesys Cloud CX BYOC certification, which means our trunks are guaranteed to interoperate correctly with the platform — not just technically compatible in theory.

Interactive Voice Response (IVR) is an automated telephony system that interacts with callers through pre-recorded voice prompts and collects input via keypad (DTMF) or voice recognition. IVR systems handle routine tasks — account inquiries, appointment confirmations, bill payments, call routing — without requiring a live agent.

Modern IVR in Genesys Cloud CX goes beyond simple menu trees. It integrates with back-end systems to retrieve real account data, uses natural language understanding to interpret spoken requests, and can escalate to an AI voicebot for more complex self-service before offering a live agent. The quality of the SIP trunk directly affects IVR performance: DTMF must be transmitted accurately, audio must be clear enough for speech recognition to function, and post-dial delay must be minimal so menu prompts play at the right time.

Jitter is the variation in the arrival time of data packets in an IP network. In VoIP and SIP trunking, jitter is one of the primary causes of audio quality degradation — when voice packets arrive unevenly, the audio sounds choppy, robotic, or has gaps. The acceptable jitter threshold for VoIP is generally under 30 milliseconds; above this level, call quality becomes noticeably impaired.

Jitter is caused by network congestion, inconsistent routing, or poor Quality of Service (QoS) configuration. A jitter buffer on the receiving device can compensate for moderate jitter by holding packets briefly and resequencing them before playback. However, a large jitter buffer introduces latency, which creates a different quality problem. The best solution is preventing jitter at the network level through proper QoS configuration and choosing a SIP provider with a low-jitter network. Star Telecom’s network is monitored continuously for jitter and latency metrics, with SLA commitments for enterprise trunks.

Key Performance Indicators (KPIs) are the quantifiable metrics used to evaluate the performance of a contact centre against its operational and business objectives. Core contact centre KPIs include: Average Handle Time (AHT), First Contact Resolution (FCR), Customer Satisfaction Score (CSAT), Net Promoter Score (NPS), Service Level (percentage of calls answered within a target time), Abandonment Rate, and Agent Occupancy.

Modern CCaaS platforms like Genesys Cloud CX provide real-time and historical KPI dashboards with drill-down capabilities. Connecting KPI data to the underlying telecom layer is important — audio quality issues, routing failures, or carrier outages will manifest as KPI degradation before they are identified as infrastructure problems.

Latency in telecommunications is the time delay between a signal being transmitted and received. In VoIP, latency is the measurable pause between when a person speaks and when the other party hears them. Acceptable one-way latency for voice calls is generally under 150 milliseconds; above 200 milliseconds, callers begin to experience noticeable conversational delay and may inadvertently interrupt each other (the “satellite call” effect).

For AI contact centre applications, latency has a compounding effect: audio must travel from caller to SIP trunk, to the Genesys media server, to the AI transcription engine, and back before a response can be generated. Each hop adds latency. Star Telecom minimizes latency through direct peering with Genesys Cloud CX data centres, avoiding unnecessary routing hops. This is one of the primary reasons specialized SIP providers outperform generic carriers for AI-powered contact centre deployments.

A Letter of Authorization (LOA) is a legal document signed by the owner of a telephone number that authorizes a new carrier to port that number away from the current carrier. The LOA is a required component of every porting request. Without a valid, correctly completed LOA, the losing carrier will reject the port request.

An LOA must include the authorized account holder’s name and signature, the account number with the current carrier, the exact telephone numbers to be ported, and the service address. Errors in any of these fields — even minor discrepancies between the LOA and the CSR (Customer Service Record) data on file with the current carrier — cause port rejections and delays. Star Telecom’s porting team provides LOA templates pre-formatted to each carrier’s requirements and reviews every submission before it is filed, significantly reducing rejection rates.

Local Number Portability (LNP) is the regulatory requirement that allows telephone subscribers to keep their existing telephone number when switching from one carrier to another, changing from a wireline to a VoIP service, or moving within a local area. In Canada, LNP is mandated by the CRTC for all local numbers.

The porting process involves coordination between the new carrier (Star Telecom), the current carrier, and the Number Portability Administration Centre (NPAC) database. Typical porting timelines for Canadian numbers range from 3–10 business days for individual numbers to 30+ days for complex enterprise migrations involving hundreds of DIDs. Star Telecom’s Handheld Porting service manages this entire process on behalf of the client.

Mean Opinion Score (MOS) is the standard measurement of perceived voice quality in telephony, expressed on a scale from 1 (bad) to 5 (excellent). A MOS of 4.0 is considered “good” quality, equivalent to a clear landline call. VoIP deployments typically target MOS scores of 3.5–4.2 for acceptable call quality.

MOS is affected by all the factors that degrade audio: latency, jitter, packet loss, codec compression, and echo. Network monitoring tools can calculate real-time MOS scores, allowing operations teams to detect degrading call quality before it reaches a level that customers complain about. Star Telecom’s network monitoring includes MOS tracking as part of enterprise trunk SLA reporting.

MPLS is a high-performance routing technique that directs network traffic using labels rather than network addresses. MPLS networks provide predictable, low-latency paths for traffic — making them well-suited for voice and video applications where packet delivery consistency matters. Many enterprises use MPLS WANs to connect branch offices to their SIP trunking gateway or CCaaS platform, ensuring voice traffic receives prioritized, consistent network treatment.

Network Address Translation (NAT) is a technique used by routers to map private internal IP addresses to a public IP address for internet communication. While NAT works transparently for most internet traffic, it frequently causes problems with SIP trunking because SIP packets embed the local IP address in the message body — the router changes the header IP but not the embedded one, causing one-way audio, registration failures, or call drops.

NAT traversal in SIP requires specific configuration: STUN, TURN, or ICE protocols on the client side, or an SBC (Session Border Controller) that handles NAT traversal on behalf of the endpoint. Star Telecom’s SIP trunking onboarding includes NAT configuration review as a standard step, preventing this common source of VoIP deployment issues.

A Network Operations Centre (NOC) is a centralised facility staffed by engineers who monitor, manage, and maintain an organization’s telecommunications network around the clock. The NOC tracks network performance, responds to alerts, investigates anomalies, and coordinates incident response.

Star Telecom operates a 24/7 NOC that monitors all enterprise SIP trunks for latency, jitter, packet loss, and availability metrics. NOC engineers receive automated alerts when trunk performance degrades below defined thresholds, enabling proactive intervention before clients experience call quality issues.

Number porting is the process of transferring existing telephone numbers from one carrier to another while keeping the numbers unchanged. Porting is regulated under CRTC guidelines in Canada, ensuring businesses can switch telecom providers without losing their established phone numbers.

The porting process requires a Letter of Authorization (LOA), Customer Service Record (CSR) data matching, coordination with the losing carrier, and a scheduled cutover window. Complex enterprise ports involving large blocks of toll-free and DID numbers require careful sequencing to avoid disruption to call routing. Star Telecom’s Handheld Porting service assigns a dedicated project manager to each enterprise migration, handling all carrier coordination on the client’s behalf. Our approach minimizes rejection rates and ensures ports complete on schedule with zero downtime.

Omnichannel in contact centre terms means providing a seamlessly integrated customer experience across all communication channels — voice, email, chat, SMS, social media, and messaging apps — so that customers can switch between channels and agents have full context of the interaction history regardless of which channel was used previously.

True omnichannel is distinguished from multichannel: a multichannel contact centre offers multiple channels separately; an omnichannel contact centre unifies them so context follows the customer. A customer who starts a chat conversation and then calls in should not have to repeat their issue — the voice agent has the chat transcript. Genesys Cloud CX is designed as a native omnichannel platform. Voice, delivered via Star Telecom’s SIP trunking, is one channel in the omnichannel stack — but it must be technically integrated with the same platform handling all other channels to deliver true context continuity.

Packet loss is the failure of one or more transmitted data packets to arrive at their destination. In VoIP, packet loss causes audio gaps, clipping, and distortion. The human ear is surprisingly sensitive to packet loss: even 1% packet loss can produce noticeable audio quality degradation; above 3%, calls become difficult; above 10%, communication breaks down.

Packet loss is caused by network congestion, hardware faults, poor routing, or insufficient bandwidth. Unlike jitter (which can be partly compensated by a jitter buffer), packet loss cannot be fully recovered — once a packet is lost, that portion of audio is gone. Preventing packet loss requires proper QoS configuration to prioritize voice traffic, adequate bandwidth provisioning, and a carrier network with reliable routing. Star Telecom’s network SLA includes packet loss commitments for enterprise trunks.

A Private Branch Exchange (PBX) is a private telephone switching system used within an organization that allows internal calls between extensions and provides a shared pool of external PSTN lines. Traditional PBXs were hardware appliances installed on-premises. Modern PBX systems are often software-defined (soft switches) or hosted in the cloud (hosted PBX or CCaaS platforms like Genesys Cloud CX).

SIP trunking connects the PBX (or CCaaS platform) to the PSTN, providing the external call paths for inbound and outbound calls. The number of SIP trunks provisioned should match the maximum number of simultaneous calls the organization expects, with additional capacity for peak periods.

The Payment Card Industry Data Security Standard (PCI DSS) is a set of security standards designed to ensure that all companies that process, store, or transmit credit card information maintain a secure environment. For contact centres that handle card payments over the phone, PCI compliance requires specific controls over how card data is captured, processed, and stored.

One of the most effective approaches to contact centre PCI compliance is DTMF masking during payment capture: when a customer enters their card number using their telephone keypad, the DTMF tones are masked in the recording and hidden from the agent screen, preventing card data from appearing in call recordings or being visible to agents. Star Telecom’s SIP trunking supports DTMF masking and PCI-compliant call recording configurations. Our partnership with RedSky also extends to secure payment capture solutions for Genesys Cloud CX deployments.

Porting is the process of transferring a telephone number from one carrier to another. It is governed by CRTC regulations in Canada, which mandate that any number can be ported as long as the account holder authorizes it and the correct documentation is provided. Porting is the mechanism by which businesses switch to Star Telecom without losing their existing phone numbers.

The porting process involves three key steps: the subscriber provides authorization via a Letter of Authorization (LOA), the CSR data is verified against the losing carrier’s records, and a port date is scheduled. Rejections occur when LOA data does not exactly match the carrier’s records. Star Telecom’s Handheld Porting service manages the entire porting workflow — CSR verification, LOA preparation, carrier coordination, and cutover scheduling — with white-glove attention to every detail.

The Public Switched Telephone Network (PSTN) is the aggregate of the world’s circuit-switched telephone networks — the traditional telephone infrastructure that has existed since the 19th century and connects virtually every telephone number on earth. Despite the shift to VoIP and cloud telephony, the PSTN remains the backbone of telephone communication globally, as all VoIP and SIP calls must ultimately interconnect with the PSTN to reach traditional phone numbers.

SIP trunking is the technology that connects a business’s IP-based phone system (PBX or CCaaS) to the PSTN. Star Telecom holds direct PSTN interconnections in Canada, which means calls from Genesys Cloud CX reach the PSTN through fewer hops — reducing latency, improving call quality, and eliminating middleman fees.

Quality of Service (QoS) refers to the set of network technologies and policies that prioritize certain types of traffic to ensure reliable performance. In a network carrying both business data and VoIP calls, QoS configuration marks voice packets as higher priority than data packets (such as file downloads or email). This ensures voice traffic receives sufficient bandwidth and low latency even when the network is congested.

Without QoS, a large file download can starve voice traffic of bandwidth, causing the same symptoms as insufficient bandwidth — jitter, latency, and packet loss. QoS is configured at the router and switch level, typically using DSCP (Differentiated Services Code Point) marking to tag voice packets. Star Telecom’s onboarding documentation includes specific QoS recommendations for routers in Genesys Cloud CX environments.

Redundancy in telecommunications means having duplicate systems, paths, or components that automatically take over when a primary element fails. For a contact centre, redundancy applies at multiple layers: the SIP trunking layer (multiple carrier paths), the network layer (redundant internet connections), the data centre layer (geo-redundant platforms), and the application layer (CCaaS platform HA).

Star Telecom builds redundancy into the SIP trunking layer by default. Every enterprise trunk deployment uses multiple carrier interconnects. If one path experiences degradation or failure, traffic automatically shifts to the backup path without manual intervention. This is not an add-on feature — it is the standard architecture. Combined with Genesys Cloud CX’s own geo-redundant infrastructure, the result is a contact centre platform with no single point of failure in the telephony stack.

A Session Border Controller (SBC) is a device or software platform that sits at the border between a private IP network and a SIP trunking provider’s network, managing and securing SIP traffic in both directions. The SBC performs several critical functions: NAT traversal, protocol normalization, security (preventing SIP-based attacks), media transcoding, and call admission control.

In enterprise SIP deployments, the SBC is one of the most important components for call quality and security. It translates between different SIP implementations (not all vendors implement the SIP standard identically), handles encryption negotiation, and prevents SIP flood attacks from reaching the internal PBX. Star Telecom is compatible with all major enterprise SBC platforms including AudioCodes, Oracle, Cisco, and Ribbon, and our technical team provides SBC configuration guides specific to Genesys Cloud CX BYOC deployments.

Session Initiation Protocol (SIP) is the application-layer signaling protocol used to initiate, manage, and terminate real-time communication sessions — including voice calls, video calls, and messaging — over IP networks. SIP is the dominant VoIP signaling standard, having largely replaced H.323 in enterprise and carrier environments.

A SIP call works in two phases: signaling and media. SIP handles the signaling — the digital handshake that sets up the call, negotiates codec parameters, and tears down the session when complete. The actual voice audio is carried separately via RTP (Real-time Transport Protocol). This separation means SIP trunking providers manage the signaling and routing layer, while audio quality depends on the RTP media path.

Genesys Cloud CX is a fully SIP-native platform. Star Telecom’s SIP trunking connects your Genesys environment to the PSTN, providing the certified, low-latency SIP connectivity required for enterprise contact centre operations.

SIP Trunking is the use of Session Initiation Protocol to deliver telephone service over an internet connection, replacing traditional physical telephone lines (ISDN, T1/PRI) with virtual trunks that carry voice calls as IP data packets. SIP trunks connect a business’s PBX or CCaaS platform to the PSTN, enabling inbound and outbound calls to any phone number worldwide.

SIP trunking typically costs 40–60% less than legacy PRI lines while providing greater flexibility — trunks can be added or removed on demand, rather than being physically constrained by the number of channels on a copper circuit. For Genesys Cloud CX deployments using BYOC, SIP trunking is the connectivity layer that determines call quality, redundancy, and total telecom cost.

Star Telecom provides certified SIP trunking purpose-built for Genesys Cloud CX. Our trunks support G.711 audio for AI feature accuracy, automated failover for 99.999% availability, Canadian DID provisioning across all area codes, and CRTC-compliant E911. Enterprise clients typically achieve 30–40% cost reduction versus incumbent carrier pricing.

A Service Level Agreement (SLA) is a contractual commitment between a service provider and a customer that defines the expected performance levels for a service, including uptime guarantees, response times, and remedies if commitments are not met. In SIP trunking, key SLA metrics include availability (uptime percentage), latency, jitter, and packet loss.

When evaluating SIP trunking providers, the SLA is one of the most important documents to review. A provider offering “99.9% uptime” allows approximately 8.7 hours of downtime per year. An enterprise contact centre requires 99.99% (52 minutes/year) or 99.999% (5 minutes/year). Star Telecom’s enterprise SIP trunking SLA includes uptime commitments, audio quality metrics, and response time commitments for critical incidents.

SMS (Short Message Service) is the text messaging protocol used by mobile networks to send and receive messages of up to 160 characters. In business contexts, SMS is used for appointment reminders, two-factor authentication, outbound notifications, and two-way customer service interactions.

For contact centres, SMS is one of the channels in an omnichannel strategy. Genesys Cloud CX supports SMS as a native interaction channel, routing inbound SMS to agents or automated flows alongside voice, email, and chat. Star Telecom provides business SMS services, including long codes (10-digit numbers) and short codes, with CRTC-compliant opt-in and opt-out management. SMS messages sent from a Star Telecom number are delivered with high-fidelity routing and carrier-grade reliability.

STIR (Secure Telephone Identity Revisited) and SHAKEN (Signature-based Handling of Asserted information using toKENs) are a pair of technical standards designed to combat cnam“>caller ID spoofing — the practice of making a call appear to come from a number that does not belong to the caller, often used in fraud and robocall campaigns.

Under STIR/SHAKEN, originating carriers digitally sign outbound calls with an attestation level: A (full attestation — the carrier knows the caller and the number belongs to them), B (partial attestation — the carrier knows the caller but cannot confirm the number), or C (gateway attestation — the call was received from another carrier). These attestation levels are passed through the network and can be displayed to call recipients or used to filter suspected robocalls.

In Canada, the CRTC mandated STIR/SHAKEN implementation beginning in 2022. Star Telecom is STIR/SHAKEN compliant on all SIP trunk origination, applying A-level attestation to verified customer numbers. This ensures your outbound calls from Genesys Cloud CX are properly attested and less likely to be filtered or flagged by recipient carriers.

Total Cost of Ownership (TCO) in the context of telecom infrastructure is the complete cost of owning and operating a telecommunications solution over a defined period, including hardware, software licenses, carrier costs, professional services, maintenance, and internal labour. TCO analysis is essential for comparing the true cost of different deployment options — not just the headline monthly rate.

For Genesys Cloud CX deployments, TCO has several components: the Genesys platform licensing (per-seat or per-interaction), SIP trunking costs (per-channel or per-minute), DID and toll-free number costs, E911 services, and professional services for setup and ongoing support. Star Telecom’s interactive TCO calculator helps Genesys prospects model their complete telecom costs under different usage scenarios, enabling accurate budget forecasting before deployment.

Clients who consolidate their SIP trunking with Star Telecom from fragmented carrier arrangements typically reduce their total telecom TCO by 30–40%, driven by volume pricing, elimination of carrier markups, and simplified billing.

Telco is shorthand for telecommunications company — any business that provides communication services including telephone, internet, and data connectivity to consumers or enterprises. In the context of CCaaS deployments, “the telco layer” refers to the SIP trunking and connectivity infrastructure that sits between the cloud contact centre platform and the public telephone network.

Choosing the right telco partner for a Genesys Cloud CX deployment is one of the most consequential decisions in the project, because the telco layer determines call quality, redundancy, cost, and AI performance. Star Telecom is a specialized telco for CCaaS environments — purpose-built for Genesys Cloud CX, not a generic carrier offering SIP as a commodity service.

Transport Layer Security (TLS) is the encryption protocol used to secure communications over a network. In SIP trunking, TLS encrypts the SIP signaling messages that set up and manage calls, preventing interception of call metadata such as caller ID, called number, and session parameters.

TLS for SIP (sometimes called SIPS) is paired with SRTP (Secure Real-time Transport Protocol) for media encryption to provide end-to-end security for both the call signaling and the voice audio. Star Telecom supports TLS/SRTP on all enterprise SIP trunks, with certificate-based mutual authentication available for high-security deployments.

Toll-free numbers (in Canada and the US: 1-800, 1-833, 1-844, 1-855, 1-866, 1-877, 1-888 prefixes) are telephone numbers for which the called party pays for the call rather than the caller. Toll-free numbers are standard for contact centre inbound lines because they remove the cost barrier for customers calling from any location.

Star Telecom provisions and manages toll-free numbers directly through our CLEC carrier status, including number provisioning, porting, and failover routing. Toll-free numbers in a Genesys Cloud CX environment can be configured with time-of-day routing, geographic routing, and automated failover to backup sites, all managed through the Star Telecom portal.

Unified Communications as a Service (UCaaS) is a cloud-based model that combines multiple enterprise communication tools — voice calling, video conferencing, instant messaging, file sharing, and presence — into a single integrated platform delivered as a subscription service. Microsoft Teams and Zoom Phone are common UCaaS platforms.

UCaaS differs from CCaaS in scope: CCaaS (like Genesys Cloud CX) is specifically optimized for customer-facing contact centre operations with ACD, IVR, omnichannel routing, and workforce management. UCaaS covers internal employee communications. Many organizations run both: UCaaS for internal collaboration and CCaaS for customer interactions, with SIP trunking connecting both to the PSTN. Star Telecom provides SIP connectivity for both UCaaS (including Microsoft Teams Direct Routing) and CCaaS deployments.

Voice over Internet Protocol (VoIP) is the technology that enables voice calls to be transmitted as digital data packets over an IP network — the internet or a private IP network — rather than through traditional circuit-switched telephone lines. VoIP is the foundational technology underlying SIP trunking, cloud PBX systems, and CCaaS platforms like Genesys Cloud CX.

Every call made through a modern business phone system is a VoIP call at some level. The voice is digitized, compressed using a codec (such as G.711 or G.729), packetized, and transmitted over IP. At the receiving end, the packets are reassembled and converted back to audio. The quality of this process — and the quality of the IP network it traverses — determines the perceived call quality.

Star Telecom’s SIP trunking delivers enterprise-grade VoIP connectivity optimized for Genesys Cloud CX, with direct PSTN interconnection, QoS-optimized routing, and G.711 audio throughout the media path.

A voicebot (also called a conversational AI agent or virtual agent) is an AI-powered system that handles voice interactions autonomously, using natural language processing (NLP) to understand what callers say and generate appropriate responses or actions — without requiring a human agent.

Modern voicebots in Genesys Cloud CX can handle complex, multi-turn conversations: verifying identity, processing transactions, escalating to a live agent with full context when needed. The performance of a voicebot is directly linked to audio quality: a voicebot’s NLP engine processes transcribed audio — if the transcription is inaccurate due to poor audio quality, the voicebot fails to understand the caller and the interaction breaks down. Star Telecom’s G.711 SIP trunking provides the high-fidelity audio that Genesys voicebots and AI models require for accurate speech recognition and effective self-service containment.

A Virtual Private Network (VPN) creates an encrypted, private tunnel over a public network connection, allowing users to securely access corporate resources remotely. For SIP trunking, VPNs are occasionally used to connect remote workers or branch offices to a central SIP gateway, though this configuration can introduce latency that affects call quality. Modern SIP endpoints typically use TLS/SRTP encryption directly rather than routing all voice traffic through a corporate VPN, which avoids the latency issue.

WebRTC is an open-source technology that enables real-time voice, video, and data communication directly through a web browser, without requiring any plugin or application installation. WebRTC is increasingly used for browser-based agent desktops in CCaaS platforms — a Genesys Cloud CX agent can handle voice calls directly through their Chrome or Edge browser using WebRTC, eliminating the need for a separate softphone application.

WebRTC in a CCaaS environment still relies on SIP trunking for connectivity to the PSTN for inbound and outbound calls. The SIP trunk terminates at the platform’s media server; the agent’s WebRTC browser session connects to the same platform. Audio quality for WebRTC calls is affected by the same SIP trunk quality metrics as traditional SIP phone calls.

Workforce Management (WFM) is the process of forecasting call volumes, scheduling agents to match anticipated demand, tracking real-time adherence, and analyzing historical patterns to continuously improve staffing efficiency. In a contact centre, accurate WFM directly impacts both cost (avoiding overstaffing) and service levels (avoiding understaffing).

Genesys Cloud CX includes native WFM capabilities integrated with the ACD, providing forecasts based on historical interaction data. Star Telecom offers WFM as a Service for organizations that want expert management of their Genesys Cloud WFM configuration, ensuring schedules are optimized and forecast accuracy improves over time.

ZRTP is a cryptographic key agreement protocol used to negotiate the encryption keys for SRTP (Secure Real-time Transport Protocol) media streams in VoIP calls. Unlike standard SRTP which relies on the SIP signaling layer to exchange keys, ZRTP performs key negotiation within the media stream itself, providing end-to-end encryption that is independent of the signaling path.

ZRTP is primarily used in peer-to-peer VoIP applications where true end-to-end encryption is required. In enterprise SIP trunking deployments, TLS/SRTP with SIP-layer key exchange is the standard approach. Star Telecom supports both configurations.