SIP phones use the internet rather than traditional copper wires to make and receive business calls. This guide explains how Session Initiation Protocol (SIP) works, how it compares with Voice over IP (VoIP), common types of phone hardware, technical requirements, and how to evaluate providers for your organization.
What Is a SIP Phone?
A SIP phone is a device or software application that transmits voice data over an internet connection using the Session Initiation Protocol. Instead of using legacy Public Switched Telephone Network (PSTN) copper wiring, SIP phones operate within a VoIP architecture to deliver wideband HD audio, lower recurring costs, and advanced call routing.
SIP phones connect to a phone system via Ethernet, Wi-Fi, or cellular data and generally fall into two categories:
- Hardware SIP phones (hard phones): Physical desktop devices and conference units that look and operate like traditional business telephones. They connect directly to IP telephony networks and remain registered with your service provider.
- Software SIP phones (softphones): Software applications installed on desktop computers, laptops, tablets, or mobile devices. Users make and receive calls using their business extension along with a headset or built-in audio hardware.


What Is SIP?
SIP stands for Session Initiation Protocol. It establishes, manages, and ends real-time voice, video, and messaging sessions between two or more devices.
Session Initiation Protocol is a communications protocol for multimedia sessions, including video and voice calls. Organizations use SIP with a phone system or service that routes communications over the internet.
SIP handles signaling, such as:
- Registering a phone with a provider
- Starting a call
- Finding the destination device
- Negotiating codecs and media settings
- Updating call status
- Ending the session
SIP does not carry the voice audio itself. It manages the session, while a separate media stream carries the audio between participants.
SIP also supports unified communications by connecting phone capabilities with video, email, instant messaging, and other communication tools.
How SIP Compares With VoIP and Related Technologies
While “VoIP” and “SIP” are used interchangeably, they are different components of an advanced communications system:
| Technology or service | What it does | Use case |
|---|---|---|
| VoIP | Transmits voice communications over an IP network instead of traditional phone lines | Business calling, conferencing, and call recording |
| SIP | Signals and manages voice, video, and messaging sessions | Connects phones, applications, PBX systems, and providers |
| SIP phone | A hardware or software endpoint that uses SIP to make and receive calls | Employee desk phones, mobile apps, and softphones |
| Hosted PBX | Places the phone system and call-routing software in the provider’s cloud | Businesses that want managed phone-system features |
| SIP trunk | Connects an organization’s IP PBX to the public telephone network | Businesses that want to keep an existing PBX or phone system |
VoIP is the technology that allows voice communication to travel over broadband internet connections. It digitizes voice data and supports features such as video conferencing and call recording. A VoIP provider can also connect calls to the public switched telephone network (PSTN)
SIP is a standardized communication protocol. It defines how devices and platforms initiate, manage, and terminate communication sessions.
In short:
- VoIP provides the technology for internet-based calling.
- SIP provides communication rules that help VoIP devices and platforms work together.
- A SIP phone is an endpoint that uses SIP within a VoIP system.
SIP Phone Features
SIP phones may resemble traditional desk phones, but they support many features beyond basic calling. Common capabilities include call holding, extension transfers, directories, conferencing, and application integration.
For IT professionals, SIP phones connect directly to VoIP service providers. This can reduce the need for separate phone servers and simplify maintenance, although the network still needs proper configuration.
Common features:
- Call forwarding: Forward calls to another number, voicemail, or multiple phones at the same time.
- On-hold music: Keep callers informed with music or custom audio messages.
- Wi-Fi calling: Make and receive calls over a Wi-Fi network.
- HD phone calls: Use wideband audio for clearer conversations.
- Auto-attendant: Greet callers and direct them to the appropriate person or department.
- Call recording: Record business calls for reference, training, or quality assurance.
- Shared call appearance: Allow multiple phones to ring for the same incoming call.
- Custom caller ID: Display a business name, department, or other approved caller ID.
- Mobile and desktop app integration: Make calls, manage contacts, and access phone features from a computer or mobile device.
- Unified communications: Combine calling with messaging, email, and video meetings.
- Call queues: Place callers in an organized queue when employees are busy.
- Voicemail and voicemail transcription: Store messages and convert them into readable text when supported by the provider.
- Power over Ethernet (PoE): A PoE VoIP phone receives power through its network cable, so it does not need a separate power cable.
- Bluetooth and USB support: Connect wireless headsets, wired headsets, or other accessories on compatible models.
- Presence and status indicators: Show whether a user is available, busy, or away.
- CRM and business application integrations: Connect call activity with customer records and other communication tools.
A business phone system should support more than voice calling when employees handle customer conversations across several channels. Useful capabilities include messaging, video meetings, call recording, conferencing, voicemail, analytics, integrations, mobile access, and administrative controls.
Types of SIP Phones
SIP phones generally fall into two categories: hard phones and softphones. SIP phones typically work with cloud-based phone systems, unlike traditional analog phones.
Hard phones: IP desk phones and conference phones
A hard phone looks and behaves like a regular telephone. These phones typically connect through Ethernet, although cordless models are also available.
Popular SIP hard-phone manufacturers include Poly, Cisco, Nextiva, and Panasonic.
Hard phones are generally classified as basic, intermediate, or advanced.
- Basic: Starter VoIP phones with a dial pad for internal and external calls. They are budget-friendly and easy to use, which makes them common among call center and contact center staff.
- Intermediate: Phones with features such as browsable directories, multiple extensions, and extra feature buttons. They are often deployed across the workforce, including employees who work from home.
- Advanced: Phones with full-color displays, multiple extensions, Wi-Fi, Bluetooth, and programmable keys. Executives, receptionists, and high-volume users often need these features.
SIP desk phones suit employees who:
- Answer calls throughout the day
- Need physical line keys or transfer buttons
- Work from a fixed desk
- Handle reception, dispatch, or shared-line duties
- Prefer a traditional handset
Their main limitation is mobility. Remote and distributed teams may prefer softphones.
Watch: SIP Hard Phone Overview and Hardware Walkthrough
Softphones: VoIP apps and contact center software
Softphones are software applications that provide phone connectivity through a desktop computer, laptop, or smartphone. After installing a VoIP app, users can connect a headset and microphone for the best calling experience.
Softphones suit employees who:
- Work remotely or across multiple locations
- Need to use a mobile or desktop device
- Switch between voice, messaging, and video
- Travel frequently
- Need access to the same business number on several devices
Watch: How Softphones Work in Unified Communications
How SIP Calls Work
SIP-based telephony uses signaling messages to establish and manage calls:
- Dialing: A user dials a phone number from a SIP endpoint.
- Call initiation: The phone sends a SIP
INVITEmessage to its registered SIP server. - Routing: The SIP server processes the request, routing the call to another extension on the network or out through the PSTN.
- Media negotiation: When the recipient answers, the devices establish a two-way audio session (via RTP) and negotiate the optimal codec and bandwidth.
- Session maintenance: The devices exchange continuous signaling messages to provide status updates (hold, transfer, mute).
- Call termination: When one party hangs up, a SIP
BYEmessage terminates the call and closes the media stream.
Bandwidth and codec requirements
A reliable deployment should also include:
- Wired Ethernet for fixed desk phones when possible
- Quality of Service (QoS) for voice traffic
- Low packet loss, latency, and jitter
- Adequate upload and download capacity
- Failover connectivity for critical phone numbers
- Network monitoring for recurring call-quality problems
Hosted PBX vs. SIP Trunking
Hosted PBX and SIP trunking both support business calling, but they solve different infrastructure needs.
A hosted PBX places the phone system in a provider’s cloud. The provider manages call routing, extensions, voicemail, auto-attendants, and other phone-system functions. Businesses connect SIP phones and software clients to the hosted platform.
SIP trunking connects an existing IP PBX to a provider’s network and the PSTN. The business keeps its PBX and usually retains more responsibility for configuration, maintenance, and internal call management.
| Business requirement | Hosted PBX | SIP trunking |
|---|---|---|
| Keep an existing IP PBX | Usually unnecessary | Yes |
| Provider manages the phone system | Yes | Partly |
| Use existing PBX hardware | Usually not required | Often possible |
| Add cloud phone features quickly | Yes | Depends on the PBX |
| Control internal PBX configuration | Limited | Greater control |
| Best fit | New or cloud-first deployments | Businesses with an existing compatible PBX |
Businesses can also use both approaches. Nextiva describes SIP trunking as a hybrid option that lets customers keep an existing PBX and phone hardware while registering the system with Nextiva. Its SIP trunking service works with Avaya, Asterisk, Cisco, 3CX, FreePBX, and Mitel systems.
Nextiva’s Trunking Portal manages phone numbers, caller ID, failover, E911, and call history. Its verified pricing models include Unlimited SIP Trunking at $14.95 per channel per month and Metered SIP Trunking at $9.95 per channel per month plus $0.008 per domestic minute. Pricing should be confirmed before purchase because plans can change.
Benefits of SIP Phones
SIP phones can help businesses reduce reliance on traditional phone lines while adding centralized controls and mobile access.
Key benefits include:
- Reduced dependence on copper phone lines and local telephone companies
- Minimal maintenance and configuration requirements
- Connectivity for office-based and remote employees
- Call encryption when supported by the provider and configuration
- Cloud-based phone-system management
- Faster access to new phone features
- Easier scaling as the organization grows
- Support for physical phones and software clients
- Integration with messaging, video, contact center, and business applications
- Number portability and virtual-number options
SIP phones still depend on the organization’s network and provider. Internet outages, poor Wi-Fi, inadequate bandwidth, and incorrect QoS settings can affect call quality. Businesses that handle critical calls should confirm failover and emergency-calling options before deployment.
Comparing SIP Phone Providers
The phone itself is only one part of a SIP deployment. The provider determines how the business manages numbers, call routing, features, support, and service continuity.
A provider comparison should cover:
| Evaluation area | What to verify |
|---|---|
| PBX compatibility | Confirm support for the current PBX, phone models, codecs, and network equipment |
| Integrations | Check CRM, help desk, messaging, video, contact center, and collaboration integrations |
| Scalability | Review how easily the business can add users, numbers, locations, and call capacity |
| Reliability | Ask about network design, uptime commitments, monitoring, and support coverage |
| Failover | Confirm how calls reroute during an outage and whether administrators can control failover numbers |
| Number management | Review porting, virtual numbers, caller ID, extensions, and number ownership |
| Bandwidth and codecs | Confirm codec support and estimate capacity for simultaneous calls |
| Mobile and desktop clients | Check support for softphones, remote work, messaging, and video |
| Security | Ask about encryption, account controls, device authentication, and administrative permissions |
| Pricing model | Compare per-user, per-channel, metered, unlimited, equipment, setup, and support charges |
| Support | Confirm availability, response times, onboarding, and configuration assistance |
The best SIP trunking provider depends on PBX compatibility, geographic coverage, number management, failover, emergency calling, codec support, and pricing. Nextiva supports existing IP PBX environments and provides controls for numbers, caller ID, failover, E911, and call history. Businesses should verify current compatibility and pricing during the evaluation.
Industry roundups also use different criteria when comparing business VoIP providers. GetVoIP’s 2026 guide lists Nextiva as “Best value and customer support,” RingCentral for collaboration and video, Zoom for video and flexibility, Dialpad for AI features, and Vonage for APIs and customization. These labels describe different use cases, so buyers should match a provider’s capabilities to their own requirements.
Top-Rated Business SIP Phones
When comparing SIP phones:
- Match devices to each employee’s needs.
- Confirm whether support and configuration are included with the SIP provider.
- Check compatibility with the provider, PBX, headset, Wi-Fi, and network equipment.
- Decide whether to purchase or lease the equipment.
- Confirm whether the device supports PoE, Bluetooth, Wi-Fi, expansion modules, and the required number of lines.
Basic SIP phones
Poly Edge E220
Price: $160

The Poly Edge E220 combines Poly’s noise-blocking technology with a modern desktop design. It is designed for straightforward use in office environments.
- Number of Lines: 4
- Display Type: Color LCD
- Screen Size: 2.8″
- PoE Available: Yes
- Bluetooth: Yes
- Wi-Fi: No
- USB port: 1
Yealink T33G
Price: $100

The Yealink SIP-T33G is an entry-level color-screen IP phone with support for four lines. It includes a backlit 320 × 240 pixel color display.
- Number of Lines: 4
- Display Type: Color
- Screen Size: 2.4″
- PoE Available: Yes
- Headset Input: RJ-9/EHS
Intermediate SIP phones
Nextiva X-815
Price: $130

The Nextiva X-815 is a six-line color SIP phone designed for call center and office workers. Its interface helps users manage several calls.
- Number of Lines: Up to 6 lines
- Display Type: Color LCD
- Screen Size: 2.8″
- PoE Available: Yes
- Gig Ethernet Ports: Yes
- Wi-Fi: Yes, USB A210 Adapter Required (sold separately)
- Speed Dialing: Yes
- Headset Input: RJ-9
- USB port: 1
Poly VVX 350
Price: $220

The Poly VVX 350 is a six-line color SIP phone designed for call center operators and small team that handles moderate-to-high call volumes.
- Number of Lines: 6 native lines (24 with pagination)
- Display Type: Color LCD
- Screen Size: 3.5″
- PoE Available: Yes
- Headset Input: RJ-9
- USB port: 2
- Bluetooth/Wi-Fi: Yes, both via dongle
- Warranty: 1 year; lifetime when rented
Advanced SIP phones
Nextiva X-885
Price: $190+

The Nextiva X-885 is a twelve-line color SIP desk phone designed for executives and receptionists. It includes programmable keys and comes preconfigured for Nextiva’s business phone system.
- Number of Lines: 12
- Display Type: Color LCD
- Screen Size: 4.3″
- PoE Available: Yes
- Gig Ethernet Ports: Yes
- USB Port: 1
- HD Voice: Yes
Cisco IP Phone 8865
Price: $350

The Cisco IP Phone 8865 is an enterprise SIP phone designed for executives and managers. It includes a color touchscreen, Bluetooth, wired and wireless headset support, integrated Wi-Fi and Ethernet ports, and support for up to 10 business phone lines.
- Number of Lines: 10
- Display Type: Color LCD
- Screen Size: 5″
- PoE Available: Yes
- Gig Ethernet Ports: Yes
- Speed Dialing: Yes
- Bluetooth: Yes
- Wi-Fi Calling: Yes
- HD Voice: Yes
- USB Port: 2
- Expansion Module Connectivity: up to 3
- Headset Input: RJ-9
Choosing a SIP Phone Service Provider
Consider the following practices when selecting a SIP phone service provider:
- Evaluate the service provider before the phone. Customer support, compatibility, network performance, and configuration assistance affect the overall experience.
- Research the network infrastructure. A cloud phone system should have multiple points of presence, or POPs, to support reliable service across the business’s operating region.
- Plan for the next 18 to 24 months. Consider how communications needs will change as the company adds employees, locations, numbers, and applications.
- Test the network. Check bandwidth, latency, jitter, packet loss, Wi-Fi coverage, and VPN behavior before deployment.
- Ask about failover. Determine how calls route during an internet, power, device, or provider outage.
- Confirm device support. Verify compatibility for existing phones, PBX systems, headsets, and conference equipment.
- Compare total cost. Include users, channels, minutes, phone equipment, setup, porting, international calling, support, and optional features.
Support quality also matters. Confirm when support is available, how quickly the provider responds, and which configuration tasks the provider handles.
Businesses that want a managed cloud phone system may evaluate hosted PBX. Businesses with a compatible on-premises PBX may compare SIP trunking services instead. A hybrid deployment can preserve existing equipment while adding cloud connectivity and provider-managed trunking features.
Getting Started With Nextiva SIP Solutions
Whether you are looking to deploy a fully managed cloud communications platform or connect your existing IP PBX using flexible SIP trunking, Nextiva delivers an enterprise-grade solution backed by 99.999% network reliability.
From pre-provisioned desk phones (like the X-815 and X-885) to intuitive softphone apps across desktop and mobile devices, Nextiva unifies crystal-clear HD voice with video meetings, team chat, CRM integrations, and intelligent call routing, all managed from a centralized dashboard.
Reach out to a communications specialist today to design the ideal SIP deployment for your organization.
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FAQs
A SIP phone is a hardware or software endpoint used to make and receive calls over an internet connection. It connects to a SIP service or phone system that manages registration and call routing.
SIP trunking is a service that connects an organization’s IP PBX to the PSTN. It allows an existing IP PBX to make and receive calls over the internet without separate traditional phone lines.
In most cases, yes. If your existing desk phones are unlocked and follow open SIP standards, they can be reprovisioned to work with your new SIP provider. Proprietary analog or legacy digital phones may require an Analog Telephone Adapter (ATA) or gateway to connect to a modern SIP platform.
A standard uncompressed SIP call using the G.711 codec uses roughly 85–100 kbps of symmetrical bandwidth (upload and download) per concurrent call. Compressed or adaptive codecs like Opus or G.729 can operate on as little as 30–40 kbps per line while maintaining solid voice quality.
SIP desk phones suit employees who work at a fixed location, handle frequent calls, or need physical line and transfer keys. Receptionists, dispatchers, and call center agents often benefit from dedicated hardware.
Softphones suit remote employees, mobile workers, and teams that need calling, messaging, and video on a computer or mobile device. Many organizations use both types so employees can select the endpoint that fits their work.
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