Today's business phone systems have a tough job. They have to provide voice communications with employees at their desks; they have to support a call center for sales, customer service, and support; and they need to connect with and through a host of other communications channels, such as fax machines, video conferencing, conference calling, mobile communications, wireless handsets, and text messaging. On top of that, they're often expected to provide more advanced functionality through software, like shared meeting collaboration, voicemail to email transcription, and call recording. And lest we forget, many businesses still need a service that will connect to public switched telephone network (PSTN).
The SIP software provides for both on-board traditional desktop services such as Caller-ID, Call Hold, Call Transfer, 3-Way Calling, and Call Waiting as well as an XML interface to allow for enhanced web based services. The XML interface allows the phone to transcend the traditional phone paradigm and become a true internet appliance. 3CX Windows VoIP Phone For VoIP Providers The 3CX VoIP Phone is a free softphone app available for Windows. Connect the app to a VoIP Provider to make calls to any mobile or landline number, just as you would from your desk phone. Installing the 3CX VoIP Phone is easy and only takes a few minutes.
Because they're working across such a multitude of channels, many of today's phone systems are adopting the moniker of Unified Communications-as-a-Service (UCaaS). These are generally cloud-based, virtual PBXes (private branch exchanges) that include at least one, usually multiple, software clients to enhance their functionality on the web, desktop, and a variety of mobile devices. UCaaS systems have a wide variety of feature sets based on the tried and true Voice-over-IP (VoIP).
A key attraction of VoIP is that it gives these systems the flexibility to work in a wide variety of environments ranging from analog desk phones to softphones piggy-backing on a cell phone. These systems can often also integrate all or part of their softphone clients into other back-office applications, like your customer relationship management (CRM) or help desk platforms. Simply picture the standard interface of such an app that suddenly sports a dial pad and some function buttons as a pop-up screen and you'll have a very basic idea of how some of this works. In addition, these cloud based systems can have a variety of phone numbers in global locations, so that your customers can have free access to your phone at little or no charge.
Before you can start considering a phone system, you need to figure out what it's going to be used for, and how much of your business will be involved. You also need to look at your existing phone system and decide whether you're going to simply keep all of it and bolt some VoIP functionality on top, retain only part of it, or replace the whole thing. Frequently, a total replacement isn't in the cards if only because some parts of your existing phone system can't be easily changed over to softphones or even desktop VoIP phones. For example, if you have a heavy manufacturing environment with outdoor activities, such as a steel fabrication yard or even a landscaping company, your old outdoor phones may be exactly what you need. You also need to decide what features of the existing phone system are required, and what features of a future phone system you feel are necessary to carry into the future.
When you're considering a new VoIP phone system for your business, it's important to include stakeholders from all of the key parts of your business in the planning and decision making process. Yes, this especially includes the IT staff and the data security folks since your voice communications will now be data. But it also needs to include folks who will be using the system to get work done, especially the work that drives revenue and engages customers. These people have invaluable insights into what's really needed versus what's simply cool and new. Plus, you'll need their input to select a phone system that will actually move your business forward as well as fit into your IT environment.
A critical part of the discussion with your IT staff will be whether your existing data network can handle the extra load that will be placed on it by the new phone system. You'll need a network that can handle more advanced network management capabilities, including tools to fight jitter and latency as well as to provide Quality of Service (QoS) and different kinds of network segmentation, especially virtual LANs (VLANs). Only tools like these can help free up your network from too much congestion, which can cause your call quality to decrease or even crash the VoIP system entirely.
On the physical side, you'll also need to plan for providing Ethernet drops to any new desktop phones you'll be placing on user desks, or even adding capacity to your Wi-Fi network should you decide to use wireless calling. For many organizations a separate network is often winds up being the preferred solution. If that's what happens in your case, you'll need a separate VoIP gateway. You'll also need security that understands voice protocols, and you'll need to have switches and routers that understand VoIP, too. By the time you've covered all those bases, a separate network is often the more effective solution rather than attempting to not only install but also integrate that much new equipment into an existing LAN.
Your IT staff will understand the basics of what needs to be done before a VoIP system can be selected and installed. That will include capacity testing on your current pipes and a thorough audit of your organization's network management capabilities to make sure they can support and secure the new flow of VoIP data. But for business-level users, selecting a phone system that will help them keep their various processes moving forward, especially the customer-facing ones, starts with understanding what VoIP really is.
Click image to expand full infographic. Image credit: Green.ch
VoIP is a method of digitizing voice signals, and then sending the digital voice information over an IP network. To accomplish this, the analog voice information is encoded using software called a codec. When it comes time to change the digital signal back to analog so that it's understandable, another codec does the job.
For a VoIP system to work, it needs a means of routing calls between users or to the outside world. In a cloud based system, a virtual PBX does that job. What that means to you is that the provider is running a large PBX operation in a data center somewhere, and slicing off a little of it to dedicate to your organization in exchange for your money. You're essentially sharing a large PBX with that provider's other customers, but because these companies use multi-tenant segmentation, your PBX will appear dedicated to you. This engine will take care of routing calls on your VoIP network.
However, for many businesses there's a need to route calls to the PSTN and other analog phones that might remain in use, too. This may mean a PSTN gateway, or even a hybrid PBX, where there's at least a small telephone switch located on-site. Note that these days, a PBX looks exactly like the other servers in your data center, except with an attached means of handling local and analog phones. Many small businesses, however, are avoiding on-premises PBXes partially due to cost savings and partially because the capabilities offered by all-cloud systems are more than advanced enough for their needs. Some virtual cloud PBXes can handle PSTN connectivity without on-site hardware requirements.
If that all is starting to sound more complex than it's worth, remember that turning your PBX into a software solution means significant opportunity for flexibility and integration that you simply can't get any other way. After all, programmers can now treat your phone like an app. Where that's taken us is to the fast-changing UCaaS paradigm. Here, traditional VoIP providers, like the ones we review as part of this review roundup, provide additional software capabilities that are all implemented and managed from a single, unified console.
While the exact features offered in any particular UCaaS solution can change radically from vendor to vendor, most include options for video conferencing, shared meeting and document collaboration tools, integrated faxing, mobile VoIP integration, and device-independent softphone clients. All of these options let customers look at communications in a whole new way, namely, in an a la carte menu-style manner where they can implement only those features their business needs and then access them any time they want and in any combination. This new approach to business communications has been growing steadily among customers over the past few years as recent research from Statista bears out.
UCaaS Projected Market Growth in US Through 2024
The Session Initiation Protocol (SIP) is a text-based protocol similar to HTML. It's the most commonly used standard for setting up and controlling phone calls in most VoIP systems. You'll run across references to SIP in most anything you do with these kinds of phone systems, especially when you're selecting the handset hardware you want to use.
While there are still a few other legacy protocols around, and a few non-SIP standards, such as H.232, SIP is what's used for the vast majority of modern VoIP phone systems. The most common use I've seen for H.232 has been in dedicated video conferencing systems. SIP, meanwhile, handles phone service, video conferencing, and several other tasks just fine, which is why its use is so widespread. Where it has trouble is with data security, but more on that in a bit.
What makes SIP so popular is not only that it's deep and flexible, but also because it was purpose-built to engage in multimedia (meaning not just audio but also video and even text) communications over TCP/IP networks. For VoIP calls, SIP can set up calls using a number of IP-related protocols, including the Stream Control Transmission Protocol (SCTP), the Transmission Control Protocol (TCP), and the User Datagram Protocol (UDP), among others. But it can also handle other functions, including session setup (initiating a call at the target endpoint—the phone you're calling), presence management (giving an indicator of whether a user is 'available,' 'away,' etc.), location management (target registration), call monitoring, and more. Despite all that capability, SIP is simple compared to other VoIP protocols primarily because it's text-based and built on a simple request/response model that's similar in many ways to both HTTP and SMTP. Yet, it's still capable of handling the most complex operations of business-grade PBXes.
SIP is built to work on a peer-to-peer (meaning endpoint to endpoint) basis. Those two points are called the 'user-agent client' and the 'user-agent server.' Remember that those are swappable, so that in SIP, the endpoint making the call is the user-agent client initiating the traffic and endpoint receiving the call is the user-agent server receiving the call.
SIP networks usually have a proxy server and a SIP gateway. The proxy sever helps lighten the functional requirements of SIP endpoints. It also acts as both client and server, but it adds functionality around call routing and policy-based management. SIP gateways handle the routing and connectivity requirements for connecting SIP calls to other networks. Typically, the advanced features of the VoIP vendors we review here are largely based on the proprietary management technology they build into their proxy servers and gateways. By offering VoIP solutions where these elements of a SIP solution are hosted in the cloud, the providers reviewed here have more flexibility in building advanced features since they have more control over deployment and reliability.
While understanding the basics of VoIP and SIP is important, setting one of these systems up will require some general network knowledge, too. For the best quality, you will need to meet a minimum upstream and downstream data throughput requirement. In addition, you'll also need to meet a minimum latency number (that is, the time between when a signal leaves a remote computer and when your system receives it), typically measured in milliseconds. It is possible to test your network connection to see if it will support a VoIP service. RingCentral offers this service from their website, other vendors like to have their service engineers do it for you.
While home VoIP systems are fairly straightforward to set up and use, a VoIP system for all but the smallest of businesses can be quite complex, In addition to have multiple users, business VoIP systems have complex feature sets that are necessary to conduct business in today's world. In addition, a business VoIP implementation must take into account the existence of the data network, even though in most cases it won't share the same infrastructure. This will mean switches and routers optimized for voice traffic, and security that's suitable for both business and VoIP.
Source: Zendesk
The products and services in this review roundup are focused on business use and because of this either provide some PBX features or serve as full-on virtual PBXes. This may mean, among other things, that they provide service to telephone sets on your employees' desks. Most also support electronic faxing in some fashion, either directly (which can be a significant challenge for some VoIP services) or by simply integrating an incoming fax with your email system. Other popular features are video conferencing and shared meeting software (so meeting attendees not only hear each other but can present presentations or documents in a shared work space).
Some form of call center capability is often available, though many times either as a different product version or simply a higher pricing tier, so be careful before you assume you'll be getting those features. These capabilities are meant to support large sales or service desk staff and their need to route and process a relatively large number of incoming customer or user calls. That means complex menu trees, an auto-attendant for routing, and service queues. You'll probably find you need interactive voice response (IVR) capabilities, and that should be backed up by support for a live operator or some other type of human intervention.
On the higher end of this space, hosted PBX providers, such as RingCentral Office, will generally offer (sometimes even require) on-premises handset hardware, like desk or cordless VoIP phones that get sent to you preconfigured to work with their service. Plug the phones in, make sure they see an internet connection, and after some self-configuration time, your VoIP service goes live auto-magically.
That situation is for fairly pristine network and business conditions, however. Companies with legacy equipment or unique business needs may need a hybrid PBX, in which a portion of the voice network remains in the analog world, while the rest is converted to cloud-based VoIP. This could happen if you occuply an older building without the necessary Ethernet infrastructure to support VoIP or if you had custom software built a long time ago that simply isn't compatible with newer phone technologies.
For SMBs, the most commonly important features you should be considering include:
One of the most exciting and clear differentiators between a cloud PBX provider and a standard telephone system is software. Your IT staff will find a host of new software tools to help monitor and manage the system. But what catches most business operators' eyes are two key capabilities that software provides: back-end integration and softphones. The latter is exactly what the name implies, a phone that's rendered entirely in software allowing any compatible device to become a phone as long as it has an internet connection, a speaker, and a microphone. More on that below.
Back-end integration with custom and third-party apps, like CRM systems, also open a whole new world for your calling data because now it can extend the phone system beyond just basic voice communication. Such integrations allows users to transfer calls to and from their mobile phone, place and receive calls from their personal phone (that appear to be coming from the business), and interact with colleagues and customers via voice and text -- all from a variety of devices. But it also allows recording and analysis of call data to measure things like customer satisfaction, understand your sales audience at a new level, and even handle customer requests and problems automatically without the customer ever being aware they never spoke to a human.
Click image to expand full infographic. Image credit: CommWorld
Most of these VoIP solutions will require stable and consistent internet connectivity at every location where wired phones are to be used. At the very least, your business phone system must have access to a business class internet link to the cloud. This should be a dedicated link through a dedicated router if you expect your phone calls to sound as if they were coming from a business and not someone's home Skype connection. But it's important to know that you will also need a router that can create a VLAN, and one that has the ability to encrypt voice traffic, and only your voice traffic. VoIP security from end to end for all calls is now a business necessity.
For larger systems, and for systems where security is critical, the old internet connection is no longer adequate. The internet doesn't do QoS, and bandwidth can be unpredictable. Network congestion can ruin a business phone call, and activities such as DNS hijacking can put your business at risk. While we all love the internet, it's not necessarily the safe place for your business voice communications. If you fall into this category, remember that while the internet runs using the IP protocol and VoIP runs over the IP protocol, that doesn't mean that VoIP must run over the internet. You can get the same software benefits of VoIP by running your voice network over dedicated lines. Sure, it will cost more, but it will also ensure crystal clear voice quality as well as the ability to implement much-improved data security.
In addition to making sure your internet service can handle your VoIP traffic, you also need to make sure your local area network (LAN) can handle it. What makes it tricky is that if you simply drop VoIP onto your network, that traffic will get processed the same as any other traffic running across your LAN, like your shared accounting application or those 20 gigabytes worth of files your assistant just stored in the cloud. The problem there is that VoIP traffic is much more sensitive to network bumps and potholes than most general office traffic. That translates to the sound breaking up or cutting out entirely, difficulty connecting over Wi-Fi, or (worst case) dropped and lost calls. Fortunately, most of the providers reviewed here have engineering staff that will contact you as part of your setup process to help your IT staffers test and optimize your network prior to deploying their solutions. That's definitely something we recommend, but there are steps you can take now to prep your LAN for VoIP and make the deployment process that much easier.
Once you've engaged with a VoIP provider, their engineers will help you determine the overall service grade of your network (look at that as your network's basic 'VoIP readiness factor') and how to tweak their service to run effectively over your infrastructure.
Another area of business VoIP support covers the growing number of mobile employees using softphones for sending and receiving calls from a laptop or mobile device. With a cloud-based PBX solution, you can have employees at different physical locations, including multiple time zones. This makes it easier to support longer business hours to cover your entire customer base. Most of the business offerings offer call routing based on the time of day and time zone.
It's also possible to switch a call from a mobile device to a desktop line or vice versa. Business products generally offer several pricing levels based on the number of lines needed, ranging from approximately $20 per line for large organizations to $35 per line for smaller groups. Even from an administrative perspective, you should be careful, however, when migrating to a new phone system. Whether you're an individual just buying a new land line or a business moving from an old-style PBX system, or even just switching to a different VoIP provider, the process should be approached carefully and only after thorough planning.
Softphones are increasing in importance in VoIP offerings to the point that for some they're the only choice. They are a critical part of UCaaS and are as common on mobile phones and tablets as they are on desktop PCs. For workers in call centers, softphones are de rigeuer because of they're the front-end window of any CRM or help desk integration. So, for example, a softphone can combine a telephone conversation with text chat and screen sharing, which means a conversation between two employees can seamlessly add more participants, handle private text chats between those participants while the call is still going on, and extend to a collaboration session in which the group shares screens, documents, and data—no prep, no reserved lines, just button clicks.
That's the basics of UCaaS, but the concept is constantly evolving to include more communication and collaboration technologies. Those capabilities also get tweaked to provide new benefits, sometimes general, sometimes aimed at specific business use cases, like call centers or help desk operations, for example. The key is integration. Voice is becoming integrated with other back-end apps.
Mobile clients are softphones optimized for a particular mobile OS and for being used in mobile situations. This means they're designed to switch easily between different cell and wireless connections on the fly. This means you can let your employees use whatever the cheapest wireless connection around them happens to be—and often that can be free. They also let your employees use your company's phone system on their own devices. Be aware, however, that there are significant security implications regarding the use of mobile softphones on employee-owned devices. While it's possible for your employees to simply download the appropriate software from their respective app store, your IT department should be involved with allowing access while also confirming that necessary security steps are taken. Also be aware that there are important reasons not to allow soft phone installations on private devices of any type because you may not be able to remove that phone client if the employee leaves the company, and because local laws may impact how much control you have over the use of the device.
With integration being at the heart of VoIP and UCaaS, you can't make a purchasing decision here without thinking about the future. On one side, think about what you'll need in 1-5 years. On the other side, consider each vendor carefully to see what they've done over the last half decade in terms of product development and keeping up with VoIP and UCaaS trends.
It's also critical that you consider the impact of mergers and acquisitions on your phone system, both from your own organization's perspective as well as your VoIP provider. Because VoIP systems turn calls into data, the whole process isn't as plug-and-play standards-based as the old-fashioned analog phone system might have been. Should your company merge with or purchase another, VoIP compatibility will become another significant IT issue.
On the phone providers' side, since this review oundup was first published, some of the products listed here now belong to other companies and some have merged into new products. If you're planning to depend on your phone system over the course of the next decade, then you should consider a vendor that's stable enough to still be around when it's time to up upgrade.
Just about anything you can picture a business needing from a phone system can be delivered by a hosted PBX solution—and generally at a cheaper price than purchasing and maintaining your own on-premises PBX. It's just a matter of selecting the right solution for your business.
Editors' note: Line2 is owned by J2 Global, the parent company of PCMag's publisher, Ziff Davis.
Pros: A network with infrastructure in Europe, the US, and Asia. Very deep feature set of VoIP and business communication capabilities. New AI integrations. An improved user interface makes RingCentral slightly more intuitive.
Cons: Hardware will cost you extra. The workflow is not as intuitive as it should be, mostly because there's no workaround to the wizards.
Bottom Line: A long-standing Editors' Choice pick, RingCentral Office is what most companies expect from a full-featured cloud PBX solution. The platform includes everything from artificial intelligence (AI) integration to capabilities like faxing, video conferencing, and custom application integration, too.
Read ReviewPros: Friendly, flexible pricing. Plenty of standard features. Solid mobile and desktop capability with easy setup.
Cons: Desktop app has stability issues and a non-intuitive design. Meetings features are difficult to find and use.
Pdf xchange pro 2012 download link. Bottom Line: 8x8 X Series replaces the previous 8x8 Virtual Office platform with a new back end that allows for expanded features and unified communications (UC) management across multiple channels.
Read ReviewPros: Promises 99.999 percent uptime with a financially backed service-level agreement (SLA). Management of Microsoft Office 365 and hosted mail possible from improved Admin console. Offers a deep and evolving feature set.
Cons: Some features aren't yet available to some customers as the company is still upgrading all of its users to the new service. Heavy focus on Microsoft for ancillary services.
Bottom Line: Intermedia was already a PCMag Editors' Choice pick, but its Unite platform makes it an even stronger choice for businesses seeking a reliable cloud PBX with a generous amount of features.
Read ReviewPros: Impressive administrative features and calling functionality. Wide range of features. Intuitive mobile apps. Better API integration than previous version.
Cons: Add-on features come at a cost. Conferencing isn't included.
Bottom Line: Vonage Business Cloud is Vonage's beefed-up Voice-over-IP (VoIP) service that targets small to midsize businesses (SMBs). Given its robust offering of features and management capabilities, you should definitely consider it, but be aware of its potential security compromises.
Read ReviewPros: Mature feature list. Support for desk phones and external legacy PBX hardware. Provides access to legacy PSTN. Equipped with softphone client and dedicated mobile clients.
Cons: Complex system that might even require outside consulting help to properly configure for some companies.
Bottom Line: This full-featured business phone system shows its maturity by filling every need most business customers might have, and by simply working well.
Read ReviewPros: Extremely fast and easy setup. Competitive cost. Covers the basics of a business phone system, including basic call routing/IVR, voicemail, and faxing.
Cons: Requires existing landlines or mobile phones. Limited features beyond the basics.
Bottom Line: Grasshopper is aimed at small to midsize businesses (SMB). In that space, it's a solid contender with a good feature set, solid mobility support, and a decent price point, too.
Read ReviewPros: Very easy to set up and manage. Scales up or down with minimal effort. Good support for both Android and iOS mobile devices.
Cons: Default settings on mobile app typically need adjusting. Some features missing and support for others is limited.
Bottom Line: Dialpad is a glimpse of what the future of small business communications will probably look like. Instead of worrying about hardware, this service focuses on software, mobility, and integrating with as many third-party applications as possible.
Read ReviewPros: Low-cost solution. Support for existing analog phones. Android and iOS apps that bring key features to mobile phones. Easy to set up and manage for those with limited telecommunications experience. Easy to use. Good online documentation. 24x7 support.
Cons: Phones must be configured by Ooma. No softphone client for Macs or PCs.
Bottom Line: Many small businesses will be attracted to Ooma Office because it's available without a contract and has most of the features you'll need out of a business phone system. Just be aware that you may outgrow its capabilities fairly quickly.
Read ReviewPros: Freshcaller is a fully featured VoIP cloud-based package intended for use in call centers or customer support that is easy to set up and can be very reasonable in cost.
Cons: Freshcaller is not a general purpose business VoIP service because it doesn't allow internal phone calls except as part of a conference feature, and it only supports soft phones on a computer or smartphone. There is no support for hardware devices otherwise.
Bottom Line: If you need a call center phone package, then Freshcaller probably has what you want, as long as you can do it with a softphone.
Read ReviewPros: Easy to set up. Low cost for a basic business phone system that includes call routing, IVR, voicemail with transcription, and faxing.
Cons: Runs only as an app on a smartphone or computer, or as a web app.
Bottom Line: Line2 may have begun as an app to let you access multiple lines on your cell phone, but today, this is a full-on cloud business phone service that squarely targets small business and does so at a very nice price. Still, you may not find every feature you want, so vet the service carefully before committing.
Read ReviewPros: Plugs into the Microsoft ecosystem for embedded application access. Allows for super-organized channel management. Nas illmatic download free. Up to 80 video feeds. Available with Office 365. Integrates with Microsoft Office.
Cons: Looks exactly like Slack. Can only be used as part of Microsoft Office 365. Audio conferencing can be expensive depending on usage. Limited support for some features with browser client.
Bottom Line: Microsoft Teams is an excellent Slack killer, providing a nice array of features, deep Office 365 integration, and a free version that's worth the download. It's a great collaboration tool if Office 365 is already part of your organization. Video conferencing is its other strong point, but there are better options for this functionality.
Read ReviewThis is a comparison of voice over IP (VoIP) software used to conduct telephone-like voice conversations across Internet Protocol (IP) based networks. For residential markets, voice over IP phone service is often cheaper than traditional public switched telephone network (PSTN) service and can remove geographic restrictions to telephone numbers, e.g., have a PSTN phone number in a New York area code ring in Tokyo.
For businesses, VoIP obviates separate voice and data pipelines, channelling both types of traffic through the IP network while giving the telephony user a range of advanced abilities.
Softphones are client devices for making and receiving voice and video calls over the IP network with the standard functions of most original telephones and usually allow integration with VoIP phones and USB phones instead of using a computer's microphone and speakers (or headset). Most softphone clients run on the open Session Initiation Protocol (SIP) supporting various codecs. Skype runs on a closed proprietary networking protocol but additional business telephone system (PBX) software can allow a SIP based telephone system to connect to the Skype network.[1]Online chat programs now also incorporate voice and video communications.
Other VoIP software applications include conferencing servers, intercom systems, virtual foreign exchange services (FXOs) and adapted telephony software which concurrently support VoIP and public switched telephone network (PSTN) like Interactive Voice Response (IVR) systems, dial in dictation, on hold and call recording servers.
Some entries below are Web-based VoIP; most are standalone Desktop applications.
Program | Operating systems | License | Costs | Protocols | Codecs | Encryption | Max conference peers | Other abilities | Latest release | ||||||||||||||||
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
AudioCodes MobilityPLUS | Windows, Android, iOS | Proprietary | ? | SIP, RTP, XMPP, STUN, ICE | G.722 wideband, G.711a, G.711u, iLBC, G.729a, SILK, GSM, VP8, H.264, Opus | TLS, SRTP | Unknown | Voice, video, IM, Group chat, content sharing, SMS and MMS over IP services, native and social network contacts integration, incoming call/IM push notifications. | 2014; 5 years ago | ||||||||||||||||
Avaya Application Server 5300 Soft Client | Windows | Proprietary | ? | SIP, RTP | Unknown | TLS, SRTP | Unknown | 2.0; 2010; 9 years ago | |||||||||||||||||
Blink | Linux, macOS, Windows | Mixed: free software versions under GNU GPLv3 + shareware versions under gplv3 with exception of including proprietary code | macOS version proprietary on App Store, free version limited to sponsored SIP provider; Windows version proprietary; Linux version open source | ICE, SIP, MSRP, RFB (VNC), XCAP | Opus, speex, G.722, GSM, iLBC, PCMU, PCMA | TLS, SRTP and ZRTP on all versions, OTR/SMP on Linux and macOS only[2] | No limit | IM, file transfer, desktop sharing, multi-party conference, wideband | Blink Qt | ||||||||||||||||
CSipSimple | Android | GPL | Free | SIP, ICE, STUN, TURN | Opus, AMR, G.711 (u-law/a-law), speex, G.722, GSM, iLBC, G.729 (need to buy a licensed plugin), iSAC, SILK (narrow-band/wide-band/ultra wide-band) | SRTP, SIP over TLS 1.0 and ZRTP | Unknown | SIP SIMPLE messaging, Support for IPv6, Integration with Android operating system with filters and rewriting rules | 1.0.2; November 2014; 4 years ago | ||||||||||||||||
Cisco IP Communicator[3] | Windows | Proprietary | Free (for Cisco SMB Partners only) | SCCP (Skinny), SIP, TFTP, HTTP (for XML Services) | G.722 wideband, G.711a, G.711u, iLBC, iSAC, G.729a, G.729ab | SRTP | Unlimited (with bridge), 2 otherwise[4] | Call Recording, Silent Monitoring, Multiple lines and directory numbers, Configurable speed dial, Calling name and number display, Call Waiting, Call Forward, Call Transfer, Three-Way Calling (conference), Call Park, Call Pickup, Redial, Hold, Barge, Callback, Extension Mobility, Message waiting indicator, iDivert, Meet Me conferencing, Group Pickup, Do Not Disturb, XML Services | 8.6.6; April 27, 2016; 3 years ago | ||||||||||||||||
CounterPath Corporation Bria | Windows, macOS, iOS, Android | Proprietary | SIP, RTP, XMPP, STUN, ICE | G.722 wideband, G.711a, G.711u, iLBC, G.729a, SILK, GSM, VP8, H.264, Opus | TLS, SRTP | 6 party voice, 3 party video | Voice and video calling, SIP and XMPP messaging support, group chat, file, image and document sharing, contact integration, incoming call/IM push notifications | 5.3.4 July 23, 2018; 14 months ago | |||||||||||||||||
Discord | macOS, Android, iOS, Windows, Linux | Proprietary | Free, Premium 'Nitro' Subscription for Additional Features. | RTP, UDP, WS, HTTPS | Opus | TLS | 5,000 soft limit[5] | IM, file sharing, in-game overlay | 02.01.2018; February 1, 2018; 19 months ago | ||||||||||||||||
Ekiga | Linux, Windows, OpenSolaris | GPL | Free | SIP, H.323, STUN, Zeroconf, XMPP, RTP | H.263, H.264/MPEG-4 AVC, Theora, iLBC, Speex, SILK, GSM, .. | No | Unknown | Video, IM, LDAP, Call Forwarding, Call Transfer, Auto-answering, PC to phone, phone to PC, Multiple accounts, USP support, Message waiting indicator, SIMPLE-based presence etc. | 4.0.1 (February 21, 2013; 6 years ago)[±][6] | ||||||||||||||||
Empathy | Linux | GPL | Free | SIP, XMPP (Jingle), ICE (STUN-TURN), Zeroconf | Unknown | No | Unknown | IM, multi-user A/V,[7] collaborative applications | 3.12.12[8](May 13, 2016; 3 years ago)[±] | ||||||||||||||||
Eyeball Chat | Windows | Proprietary | Free | SIP, STUN, ICE, XMPP | Unknown | Yes | 5[9] | IM, Conferencing, Voice, Video and SIMPLE based presence | Windows 3.2; 2009; 10 years ago | ||||||||||||||||
eyeBeam | Windows | Proprietary | ? | SIP-SIMPLE | Unknown | TLS, SRTP | 6 party audio; 3 party video | Voice and Video calling; IM; Import Microsoft Outlook Contacts; USB Support; Call Recording and Conferencing | 1.5.20.1; March 2010; 9 years ago | ||||||||||||||||
FaceTime | iOS, macOS | Proprietary | Free | SIP, IETF, Signaling protocol for VoIP, STUN, TURN and ICE[disambiguation needed] IETF, technologies for traversing firewalls and NAT[disambiguation needed] | H.264 Video, AAC-ELD[disambiguation needed] Audio, H.263 and VP8 | RTP, SRTPIETF standards for delivering real-time and encrypted media streams for VoIP. | 20 | Video, voice, conferencing, with additional tools available as 'Services'. | Template:Latest stable software release/FaceTime | ||||||||||||||||
Google+ Hangouts | Linux, Windows, macOS, Android, iOS | Proprietary (using libjingle) | Free | XMPP | H.264/SVC, H.264/AVC, H.263 and VP8 | SRTP | 10 | Video, chat, screen sharing, with additional tools available as 'Hangout Apps'. | Google Chrome Web Store extension 2013.626.1614.1 (July 9, 2013; 6 years ago)[±][10] | ||||||||||||||||
IBM Sametime | Linux, macOS, Windows, mobile | Proprietary | ? | SIP-SIMPLE, T.120 | H.323 | TLS | Unknown | IM, File transfer, Voice, Presence, Server stored contact list, HTTP tunneling, plugins, embedable in Lotus Notes[11] | 8.5.2; May 8, 2011; 8 years ago | ||||||||||||||||
iCall | Linux, macOS, Windows, iOS, Android | Proprietary | Free | SIP, AIM, ICQ, XMPP | Speex, CELT, WebM | TLS, ZRTP | No limit | Video, file transfer, PC to phone, phone to PC, IM (MSN, AIM, ICQ, Yahoo!, XMPP, Google Talk), Voicemail | 7.1.522 | ||||||||||||||||
Jami by Savoir-faire Linux | Android, FreeBSD, iOS, iPhone, Linux, Microsoft Windows, OS X[12] | GPL3 | Free | SIP, RTP, STUN per account, SRV, DHT, P2P | Audio: Opus, Speex, G.722, G.711, GSM, VP8, G.729, iLBC. Video: H.264, H.263, VP8, MPEG-4 | Voice encryption (SRTP with SDES or ZRTP) and signaling encryption (TLS), multiple realms authentication mechanism | No limit | Blockchain ID-management, Gnome-KDE client, address book, multiple accounts, unlimited call number, call transfer, call hold-unhold, call recording, multi-way conferencing |
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Jitsi | Linux, macOS, Windows (all java supported). ExperimentalAndroid builds are also available.[17] | Apache | Free | SIP-SIMPLE, XMPP-JingleSTUNICE, TURN | SILK, G.722, Speex, Opus, G.711 (PCMU/PCMA), iLBC, GSM, G.729, H.264, H.263, VP8 | ZRTP, SRTP, OTR, TLS | Unknown | Text messaging, audio-video telephony, IPv6 (often broken,[18] P2P not supported[19]), call recording, 64-bit | 2.10 (build.5550) (February 5, 2017; 2 years ago)[±] | ||||||||||||||||
KPhone | Linux (KDE) | GPL | Free | SIP, STUN, NAPTR-SRV | Unknown | SRTP | Unknown | Video, voice, IM, external Sessions, IPv6 support for UDP | 1.2 (November 2008) | ||||||||||||||||
Librestream Onsight Connect | Windows, Android, iOS | Proprietary | ? | SIP, RTP, STUN | G.711, H.264, MPEG, and others | TLS, SRTP | Unknown | Voice, video, conferencing, image sharing, incoming call/IM push notifications. | February 27, 2018; 19 months ago | ||||||||||||||||
Linphone | Linux, Windows, macOS, Android, iPhone, BlackBerry | GPL | Free | SIP | Speex, Opus, G711, GSM, G.722, VP8 (WebM), H263, MPEG4, Theora and H264 (plugin) | TLS, SRTP, ZRTP | Unknown | Video, IM, STUN, IPv6 (disables IPv4 support when enabled, P2P supported only by version 3.5.1-2) | 3.11.1; March 10, 2017; 2 years ago | ||||||||||||||||
Messages | macOS | Proprietary | Free, only macOS and iOS | H263, H264 | Unknown | Unknown | Integrated, PBX independent | 7.0; July 25, 2012; 7 years ago | |||||||||||||||||
MicroSIP | Windows | GPL | Free | SIP, STUN, ICE, SIMPLE | Speex, iLBC, GSM, G.711, G.722, G.729, SILK, Linear PCM | TLS, SRTP | Unknown | Video, voice, IM and Presence | 3.19.15 May 2019; 4 months ago | ||||||||||||||||
Mirial Softphone (Mirial s.u.r.l.) | Windows 2000-XP-2003-Vista-7 (including 64-bit versions), macOS (x86) | Proprietary | not free | SIP, RTSP | H.323 | DTLS-SRTP | Unknown | H.264 Full-HD 1080p video rx/tx, Two independent lines supporting Call Control and 3-Party videoconference in Continuous Presence, G.722.1/C wideband audio, Call recording/export, DV/HDMI/Component capture, Presentation (H.239, RFC-4796), Encryption, Far End Camera Control, GPU accel (D3D and OpenGL) | 7.0.24; May 26, 2010; 9 years ago, discontinued | ||||||||||||||||
Mumble | Linux, macOS, iOS, Windows, Android | New BSD license | Free | ICE | CELT, Speex, Opus | TLS and OCB-AES128 | No max (limited only by server bandwidth and memory) | Chat with (limited) embedded HTML, Automatic Gain Control, very low latency, Access Control Lists for user management, Customizable In-Game Overlay for OpenGL and DirectX, Directional Audio, Plugin Support, Nested Channels, Echo cancellation for headset free use, Global Public Server List, Logitech G15 support, Push-To-Talk and Voice-Activation | 1.2.19; January 27, 2017; 2 years ago | ||||||||||||||||
Nymgo | Windows, Android, iOS | Proprietary | Free | SIP, RTP and RTCP | Unknown | Yes | No | Address Book integration, Call recording/export, Mute, On Hold, Caller ID definition | 4.2.9; March 2013; 6 years ago | ||||||||||||||||
oovoo | macOS, iOS, Windows, Android | Proprietary | Free | SIP, RTP and RTCP | Unknown | Yes | 12 | Address Book integration, Call recording/export, Mute, On Hold, Caller ID definition | 4.2.9; March 2013; 6 years ago | ||||||||||||||||
Phoner | Windows | Proprietary | Free | SIP, TAPI, CAPI | G.711a, G.711u, G.722, G.726, G.729, GSM, iLBC, speex, Opus | TLS, SRTP, ZRTP | 8 | Conferencing, call redirection, call recording | 3.21 (18 January 2019; 8 months ago)[±][20] | ||||||||||||||||
PhonerLite | Windows | Proprietary | Free | SIP | G.711a, G.711u, G.722, G.726, G.729, GSM, iLBC, speex, Opus | TLS, SRTP, ZRTP | 8 | Conferencing, call redirection, call recording | 2.72 (23 May 2019; 4 months ago)[±][21] | ||||||||||||||||
QuteCom | Linux, macOS, Windows XP-2000 | GPL | Free | SIP | G.711, G.729, iLBC, AMR-NB, G.722, Speex, AMR-WB (G.722.2), H.263, H.263+, H.264, Dirac[22] | SRTP, but key exchange via Everbee key Exchange which is not a Standard | Unknown | Video, IM (MSN, AIM, ICQ, Yahoo!, XMPP, Google Talk), voicemail, wengo to phone, conferencing. | 2.2.1 (22 June 2011; 8 years ago[23])[±] | ||||||||||||||||
Roger Wilco GameSpy | Windows | Proprietary | ? | Proprietary | ? | ? | ? | ? | 2001; 18 years ago | ||||||||||||||||
Signal | Linux, macOS, Windows, Android, iPhone | {{open source, The clients are published under the GPLv3 license, while the server code is published under the AGPLv3 license}} | Free | <to be completed> | <to be completed> | <to be completed> | <to be completed> | Signal also allows users to send text messages, files, voice notes, pictures, GIFs, and video messages over a Wi-Fi or data connection to other Signal users on iOS, Android and a desktop app. The app also supports group messaging, read receipts and typing indicators, both of which can be disabled. |
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Skype | Linux(with limited functionality),[27]macOS, Windows 2000-XP-Vista-7-Mobile (unsupported), BREW, Windows Phone , Android, iPhone, PSP | Proprietary | Free | Proprietary P2P protocol [a] | SILK | TLS | 25 starting with version 3.6.0.216. 10 with 2.x | Conferencing, video, file transfer, voicemail, Skype to phone, phone to Skype, additional P2P extensions (games, whiteboard, etc..); depending on platform. |
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TeamSpeak | Linux, Windows, macOS, FreeBSD, Android, iOS | Proprietary | Free | Unknown | CELT, Speex, Opus[32] | Yes | 32 unlicensed, 512 with non-profit license, 2000 | Simultaneous server conferencing with tabs, 3D sound effects, scalable permissions system, firewall friendly file transfers, in-game overlay for DirectX & OpenGL games, global public server list, plugin system | 3.3 | ||||||||||||||||
TeamTalk | Linux, Windows, macOS, Android, iOS, Raspbian | Proprietary | Free | Proprietary | Opus, Speex, WebM | No | 1000 | Video, file sharing, desktop sharing, stream media files (MP3, AVI) | 5.3.3; November 2018; 10 months ago | ||||||||||||||||
TeamViewer | Linux, Windows, macOS, iOS, Android[33] | Proprietary | Free (personal use only) | Unknown | Unknown | AES256 | Unknown | Unknown |
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Telephone | macOS 10.10.2 | BSD | Free | SIP, STUN, ICE | Unknown | No | Unknown | Address Book integration | 1.1.4; March 6, 2012; 7 years ago | ||||||||||||||||
Toktumi Unlimited, Line2 Pro | Windows XP-Vista-7, macOS, iOS, Android | Proprietary | Free | Proprietary with SIP core | Unknown | Unknown | 20 | Conferencing, voicemail, caller ID, call-waiting, address book integration; auto-attendant, call-forwarding | Windows; August 2010; 9 years ago; macOS; October 2010; 8 years ago | ||||||||||||||||
Tox | Linux, macOS, Windows, Android, FreeBSD | GPL | Free | Tox, VP8 | Opus, | NaCl | Unknown | Voice, video, instant messaging, file transfers | Unknown | ||||||||||||||||
Tru App | Windows 2000-XP-Vista-7, macOS, Linux iOS, Android, Symbian, BlackBerry OS, | Proprietary | Free | SIP, XMPP | Unknown | Unknown | Unknown | Chat, file transfer, voicemail, inbound numbers, integration with GTalk, Microsoft Live, Skype | |||||||||||||||||
Tuenti | Android, iPhone, Windows Phone | Proprietary | Free | WebRTC, SIP, XMPP | iLBC, Opus | Yes | ? | Voice, video, Instant messaging, group chat, photo and video sharing, SMS and MMS, native and social network contacts integration, incoming call/IM push notifications. | |||||||||||||||||
Twinkle | Linux | GPL | Free | SIP | G.711 A-law μ-law, G.726, GSM, iLBC, Speex narrow wide ultrawide | SRTP, ZRTP | 3 | Conferencing, chat, file transfer, Firefox integration, call redirection, voicemail, support of VoIP-to-Phone services | 1.10.2 (February 14, 2019; 7 months ago)[±] | ||||||||||||||||
Ventrilo | macOS, Windows, iOS, Android | Proprietary | Free | Unknown | Unknown | No | 8 | Conferencing, chat, text-to-speech | 3.0.8 | ||||||||||||||||
Viber | Linux,[b]macOS,[b]Windows,[b]Android, Bada, BlackBerry OS, iOS, Series 40, Symbian, Windows Phone | Proprietary | ? | Unknown | Unknown | Yes[44] | Unknown | Varies by platform: Text, picture and video messaging on all, voice calling only on iPhone, Android and Microsoft's Windows Phone |
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Vonage | Linux, macOS, Windows, Android, iOS | Proprietary | not free | SIP, STUN, ICE, TURN | Audio: Opus, G.711 Video: VP8 | SRTP | 50 | VoIP , video, instant messaging, presence (SIP SIMPLE), PSTN (inbound and outbound), call waiting, call hold, call forwarding, voicemail, message-waiting indication, 3-way conferencing, contacts integration, receptionist console, video, group chat, content sharing, SMS over IP services, native and social network contacts integration, incoming call/IM push notifications, fax, file sharing, screen sharing (desktop only), number programmability. | |||||||||||||||||
Wire | Linux, Windows, macOS, iOS, Android, Web | GPLv3 | not free | ? | Template:Opus (audio), VP8 (video) | DTLS, SRTP[49] | 10[50] | End-to-end encryption by default for everything, instant messaging, video call, video group call, file sharing, GIF sharing, push to talk, edit message, delete message (on both side), timed messages, doodling, identity verification, screen sharing (desktop only) | ? | ||||||||||||||||
X-Lite | macOS, Windows | Proprietary | Free | SIP, STUN, ICE, TURN | H.263, H.263+, G.711, iLBC, Speex | No | 3 | VoIP over WiFi, 3G, 4G, video, instant messaging, presence (SIP SIMPLE), call waiting, call hold, call forwarding, voice mail, message waiting indication, 3 way conferencing, contacts integration, background noise reduction (BNR), automatic gain control (AGC) | 5.3.3 July 10, 2018; 14 months ago | ||||||||||||||||
Yahoo! Messenger | Classic Mac OS (8, 9), macOS, Windows, (Linux, FreeBSD version VoIP incapable) | Proprietary | Free | SIP (using TLS) and RTP (media) | Unknown | Unknown | Unknown | Video, file transfer, PC to phone, phone to PC | Windows: 11.5.0.228; May 31, 2012; 7 years ago Mac: 3.0.1; July 20, 2011; 8 years ago Linux: 1.0.6; September 2003; 16 years ago SunOS 5.7: 0.99.17-1; September 2003; 16 years ago Solaris 8: 1.0.4; September 2003; 16 years ago FreeBSD 4-5: 1.0.4; September 2003; 16 years ago iOS: 2.2.6; July 12, 2012; 7 years ago | ||||||||||||||||
Yate Client | Linux, macOS, Windows | GPL | Free | SIP, IAX, XMPP, H.323 | G.711a, G.711u, GSM 06.10, iLBC, Speex, G.723, G.726, G.728, G.729 | SRTP, maybe ZRTP? | Unknown | 6.0.0; September 2017; 2 years ago | |||||||||||||||||
Zfone | Linux, macOS, Windows | Viewable source | Includes time bomb provision | SIP, RTP | Unknown | SRTP, ZRTP | Unknown | Beta 2008-09-04 (Linux 0.9.224), (macOS 0.9.246), (Windows 0.9.206) |
For mobile VoIP clients:
Program | Operating systems | License | Open source | Protocols | Codecs | Encryption | Other abilities | Latest release | ||||||
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Acrobits Softphone | Android v2.2+, iOS v7.0+ | Proprietary | No | SIP | G.711, G.722, iLBC, GSM, G.729 | TLS, SRTP, ZRTP | VOIP over Wi-Fi or 3G iOS only: push notifications, video, number rewriting, address book matching, sms for betamax providers and pennytel | Acrobits Softphone: 3.20 (Android), 5.2 (iOS) | ||||||
CounterPath Corporation Bria | Android v4.4+, iOS v10+ | Proprietary | No | SIP | G.722 wideband, G.711a, G.711u, iLBC, G.729a, SILK, GSM, VP8, H.264, Opus | TLS, SRTP, ZRTP | Voice and video calling, SIP and XMPP messaging support, group chat, file, image and document sharing, contact integration, incoming call/IM push notifications. | 5.3.4 July 23, 2018; 14 months ago | ||||||
CSipSimple | Android | GPL | Yes | SIP | Opus, AMR-WB, G.722, iSAC, iLBC, Speex, Silk, Codec2, G.726, G711 (PCMA & PCMU), AMR, GSM | SRTP, SIP over TLS 1.0 and ZRTP | ? | 1.02.02 | ||||||
Google Duo | iOS, Android | Proprietary, freeware | No | WebRTC | ? | ? | One-to-one voice and video calling. |
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iCall Mobile | iOS v4.3+ | Proprietary, freeware | No | SIPAIMICQXMPP, Facebook, Yahoo! Messenger, Windows Live | Speex, CELT, WebM | TLS, ZRTP | VoIP over Wi-Fi or 3G, SMS, voicemail | 2.0.1; April 2012; 7 years ago | ||||||
Jajah Mobile Web | Symbian, Windows Mobile (version unknown) | Proprietary | No | ? | ? | ? | web based service, Call back,no download[53] | ? | ||||||
JusTex by Juphoon | Android, iOS v5.1+ | Proprietary | No | SIP | PCMA, PCMU, G.722, iLBC, iSAC, Opus, H.264, VP8, H.263 | TLS, TCP, UDP, SRTP | JusTex Softphone over Wi-Fi or 3G, HD voice and video call, Multiparty call, conference. | 3.0.1 (iOS); July 13, 2014; 5 years ago; 3.0 (Android); July 9, 2014; 5 years ago | ||||||
Line | Android, iOS | Proprietary | No | ? | ? | ? | ? | |||||||
Line2 | Android, iOS | Proprietary, freeware | No | Proprietary with SIP core | ? | ? | Tri-mode calling (cellular, 3G/4G data, Wi-Fi), SMS over IP, visual voicemail, 20-person conference calls, auto-attendant, call-forwarding | 1.0.4 Android, 3.2.1 iOS | ||||||
Media5-fone | iOS, Android | Proprietary | No | SIP | PCMA, PCMU, G.722*, iLBC, iSAC*, G.729* | TLS*, SRTP* | Wi-Fi and 3G/4G data, second call*, conference calls*, HD Voice, Bluetooth* (* Additional fees) | ? | ||||||
Signal by Signal Messenger | iOS, Android | GPLv3 | Yes | WebRTC[54] | Opus[54] | TLS, Signal Protocol[54] | End-to-end encryption by default for everything. CallKit and location privacy,[55] one-to-one and group messaging, video calling,[55] image/video sharing, timed messages, identity verification, screenshot blocking. Android only: SMS/MMS messaging, doodling,[56] GIF sharing.[57] |
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Sipdroid | Android | GPL | Yes | SIP | ? | ? | Uses Wi-Fi, 3G or EDGE | 2.7 | ||||||
Tango by TangoME Inc. | iOS, Android, Microsoft Windows, macOS, Windows Phone | Proprietary, freeware | ? | ? | ? | ? | VoIP, Wi-Fi out & in, SMS over IP, call-through & call-back, instant messaging, videoconferencing | ? | ||||||
Truphone | Nokia-Symbian, iOS, Android, BlackBerry | Proprietary, freeware | No | SIP | ? | ? | VoIP, Wi-Fi out & in, SMS over IP, call-through & call-back, connection management, provisioning | Symbian 4.0, iOS 1.11.1 | ||||||
Vopium | Symbian, Java ME, Android, BlackBerry RIM, iOS, Windows Mobile 2003 SE and higher | Proprietary, freeware | No | SIP, MSN, Skype, Yahoo, AOL, ICQ, Google Talk, Facebook & Twitter | GSM | ? | Wi-Fi VoIP, GSM call-through, SMS over IP, least cost routing, synchronising-backup mobile contacts and calendar | 2.0 | ||||||
Android, BlackBerry, iOS, Symbian, Windows Phone | Proprietary, freeware | No | ? | ? | ECDH, SRTP[61] | VoIP and instant messaging over Wi-Fi or a data connection. |
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Windows Mobile 6 | Windows Mobile 6 Professional/Standard | Proprietary | No | SIP to publicly and non-publicly routable servers | ? | ? | 6.1 | |||||||
X-PRO for Pocket PC | Windows Mobile 2003 (earlier versions support Windows PocketPC 2002) | Proprietary, discontinued in April 2007[64] | No | SIP | ? | ? | Supported devices: AudioVox Maestro, Compaq/HP iPAQ: 365x, 37xx, 385x, 395x, hx4700, Dell Axim X51, X51v, Toshiba e550, e570. iPAQ 545x and Toshiba e7xx are not supported.[65] | 2.2 | ||||||
Yuilop | Android, iOS, Windows Phone, BlackBerry OS | Proprietary, freeware | No | XMPP, RTP | iLBC | TLS | VoIP over 3G-4G-LTE and Wi-fi, SMS, group chat, photo sharing, Location sharing, virtual phone numbers | iOS 2.4 (November 26, 2014; 4 years ago[66])[±] Android 2.0.2 (February 12, 2014; 5 years ago)[±] |
Program | Operating systems | License | Protocols, based on, compatible with | Encryption | Other abilities | Key and target markets | Latest release |
---|---|---|---|---|---|---|---|
Tapioca | Linux | GPL | Telepathy (software) | No | 0.3.9; June 12, 2006; 13 years ago | ||
Telepathy, Farstream | Linux, macOS, Windows | LGPL | SIP, XMPP (Jingle), ICE (STUN/TURN), UPnP | No | Multi-user A/V conferencing, IM, collaborative applications | Mobile devices (Maemo, Meego), Linux desktop or embedded | spec 0.27.2; September 24, 2013; 6 years ago |
OPAL | Windows, Linux (including embedded variants), macOS | MPL | SIP, H.323, IAX2, CAPI, VXML | Unknown | Multi-user A/V conferencing, IM, IVR | Softphones, softswitches, telephony application servers | 3.14.3; October 10, 2014; 4 years ago |
GNU oSIP | Linux, Windows, macOS, Android, iPhone, BlackBerry | LGPL | SIP, SDP | Unknown | Multi-user A/V conferencing, IM, IVR | Softphones, embedded and mobile devices, telephony application servers | 4.1.0; December 18, 2013; 5 years ago |
Program | Operating systems | License | Protocols, based on, compatible with | Encryption | Other abilities | Key and target markets | Latest release |
---|---|---|---|---|---|---|---|
3CX Phone System | Windows, Linux | Proprietary | SIP | TLS, SRTP | Voice and video IP telephony and conferencing, voice mail and instant messaging | < 50,000 users | 16.2 ; July 5, 2019; 2 months ago |
AS5300 | Linux, Windows Server 2003 | Proprietary | SIP, UNIStim, MLPP | SSL, TLS, SRTP, SDESC | Voice and Video IP telephony, Voice and Video conferencing, voice mail and instant messaging | 1,000 - 25,000 users | 1.0; January 2008; 11 years ago |
AskoziaPBX | no additional OS required (Linux based) | GPL, Open core | SIP, H.323, IAX, SCCP | No | ISDN, analog, Voicemail, Conferencing, MOH, ACD, IVR, Call forwarding, Call recording | SMEs up to 50 users | 2.2.2; December 2012; 6 years ago |
Asterisk PBX | Linux, BSD, macOS, Solaris | GPL, optional: Proprietary[68] | SIP, H.323, IAX, MGCP, VOFR, XMPP, Google Talk, TDM | TLS, SRTP | VoIP Gateway, voicemail, basic accounting (can be expanded by interface with ODBC-compliant database), billing, conferencing, hot-desking, IVR trees with conditional logic, call queuing, automated call distribution | Enthusiasts, developers, enterprise users (capacity dependent on server design, scalable across multiple servers) | 16.5.0 (25 July 2019; 2 months ago[69])[±] 15.7.3 (11 July 2019; 2 months ago[70])[±]13.28.0 (25 July 2019; 2 months ago[71])[±] |
Brekeke SIP Server | Linux, Windows | Proprietary | SIP | TLS, SRTP | SIP Registrar, SIP Proxy | SIP Service Providers, VoIP service providers, Healthcare, Security | 3.9.1.3; December 21, 2018; 9 months ago |
Brekeke PBX | Linux, Windows | Proprietary | SIP | TLS, SRTP | Voice and Video IP telephony, Voice and Video conferencing, and voice mail | Hosted service providers, Mid-large enterprise | 3.9.1.4; December 27, 2018; 9 months ago |
CallMax Softswitch | Linux | Proprietary | SIP, H.323 | SSL, TLS, HTTPS | Integrated billing, IP PBX Platform, Calling card platform, Callshop module, Customer Web Portal, Retail SMS Platform. | SIP Service Providers, Residential & Business VoIP providers, Corporate Clients, Other Class 5 softswitch users | 3.6; October 2015; 3 years ago |
CommuniGate Pro | Linux, BSD, macOS, Windows, Solaris, HPUX, AIX | Proprietary | SIP, XIMSS Protocol, XMPP, WebRTC | SSL, TLS, SRTP | SIP Registrar/Proxy, Authentication, Diameter, RADIUS, ENUM, many others | Carriers, enterprises, MNOs, ISPs, SaaS providers | 6.1.11; June 2016; 3 years ago |
Dial-Gate Softswitch PBX | Linux, Windows | Proprietary | SIP | TLS, SRTP | Billing server, real-time account and line monitoring, web-based user interface | Softswitch users, service providers | 4.3; June 2014; 5 years ago |
Dial-Office IP-PBX | Linux, Windows | Proprietary | SIP | TLS, SRTP | Unified Communications, Conferencing, remote worker support and voice mail | Small businesses, Mid-large enterprises | 4.1; December 2013; 5 years ago |
Elastix | Linux | GPL | SIP, IAX, H323, XMPP | TLS, SRTP | Unified communication server that also supports chat, mail and fax. | Capacity dependent on server design, scalable across multiple servers | 2.5.0 (stable), 3.0.0 (stable), 4.0.0 (stable); February 10, 2016; 3 years ago |
FreeSWITCH | Linux, BSD, macOS, Solaris, Windows | Mozilla Public License | SIP, NAT-PMP, STUN, SIMPLE, XMPP, Google Talk (Jingle), IAX, H.323, MRCP, RSS, Skype | TLS, SRTP, ZRTP | Recording, Voicemail, Conferencing, RADIUS, ENUM, IM Proxy, Streaming, Media gateway, Soft-PBX, IVR (modular) | Large soft-switch users, home PBX users, softphone users | 1.8.7 (July 2, 2019; 2 months ago)[±] |
FreePBX | Linux, BSD, Solaris | GPL | SIP, IAX, H323, XMPP | TLS, SRTP | Complete PABX Service, based on Asterisk and PHP 5.6; provides a full replacement for a legacy non-VoIP phone system; under current and active development | Scales from Raspberry PI (3 users) to multiple parallel clusters (10K+ simultaneous calls) | 14.0; August 2017; 2 years ago |
GNU Gatekeeper | Linux, FreeBSD, macOS, Windows XP-2000-Vista-7 | GPL | H.323 | H.235 | H.460.18 firewall traversal, routing, accounting | video conferencing, VoIP carriers large and small | 3.7; August 15, 2014; 5 years ago |
HERO Hosted PBX | Linux, Windows | Proprietary | SIP | TLS, SRTP | Unified Communications, billing server, cloud-based management and web interface | Mid-large enterprises, VoIP carriers and service providers, telecom operators | 4.3; December 2013; 5 years ago |
Kerio Operator | Linux, VM | Proprietary | SIP | SSL, TLS, SRTP | integrated firewall, Auto attendant, Call queues, Conference calling, Call forwarding, pickup, parking, recording, Click to Call, Video calling, Fax support, Paging, Salesforce.com and CRM integration, Voicemail to email, complementary desktop Softphone app | SME | 2.5.2; November 15, 2016; 2 years ago |
MediaCore SBC | Linux | Proprietary | H.323, SIP, SMPP | SSL, TLS, HTTPS | Dynamic routing mechanism including LCR, Jurisdictional routing support, A-number based routing for EU-based providers), integrated billing, SIP-H.323 protocol converter, Transcoding - codec converter module, Guardian - revenue assurance module, SBC functionality, full and media proxy, Carrier SMS support | Transit VoIP services providers, VoIP wholesale carriers, VoIP termination providers, SMS carriers | 4.6.1; May 2017; 2 years ago |
Murmur | Linux, BSD, macOS, Windows | BSD, GPL | CELT, Speex, Opus | TLS | Chat with (limited) embedded HTML, ACLs for user management, Customizable In-Game Overlay, Directional Audio, Plugin Support, Nested Channels | Individuals to Small and medium enterprise (25-5000 users) | 1.2.17; September 24, 2016; 3 years ago |
Mysipswitch | Linux | BSD | SIP, Ajax | SSL | SIP proxy server which allows the use of multiple SIP accounts with a single SIP login | Individuals | August 2007; 12 years ago |
Objectworld UC Server | Windows XP-2003-2008 | Proprietary | SIP | No | IP PBX, personal assistants, IVR, automated phone provisioning, fax server, unified messaging, Outlook, Exchange and Lotus Domino-Notes integration, conferencing, outbound dialing | Small and medium enterprise (25-2000 users) | 4.4.2; May 2009; 10 years ago |
Kamailio, OpenSIPS (formerly named OpenSER) | Linux, BSD, Solaris | GPL | SIP, XMPP | TLS, SRTP | SIP registrar-proxy, authentication, Diameter, RADIUS, ENUM, least-cost-routing, many others | SIP Service Providers | 4.0.1 (25 April 2013; 6 years ago)[±] |
Pbxnsip | Linux, BSD, macOS, Windows | Proprietary | SIP | SRTP | IP PBX, presence indication, IVR, automated phone provisioning, fax server, unified messaging, Outlook, Exchange integration, conferencing, outbound dialing | Small and medium enterprise (25-256 users) | 3.3.1.3177; April 2009; 10 years ago |
SIP Express Router (SER) | Linux, BSD, Solaris | GPL | SIP | No | SIP Registrar/Proxy, Authentication, Diameter, RADIUS, ENUM, many others | SIP Service Providers | 2.0.0 Ottendorf |
sipXecs IP PBX | Linux | AGPL | Native SIP call control, XMPP | TLS, SRTP | Full redundancy (HA), instant messaging, voicemail, user portal, admin GUI, plug & play management including phones and gateways, fully featured | Enterprises between 10 and 10,000 users, multi-site | 14.04.2; July 2014; 5 years ago |
vzRoom | Windows | Proprietary | SIP | SSL, TLS, AES | Instant messaging-chat, VoIP, video, sharing (desktop, video, file), whiteboard, scheduler, recording | Individual to small and medium enterprise (2-1,000 users) | 0.8.8.735; November 2010; 8 years ago |
Yate | BSD, Linux, macOS, Windows | GPL | SIP, IAX, H.323, ISDN, XMPP (Jabber), Jingle (Google Talk), MGCP, SS7 over IP, Cisco SLT (Signalling Link Transport) (SS7 MTP2 backhaul over IP), SCTP, SCCP, TCAP, MAPCAMEL | SSL, TLS, SRTP | Voice, video, file transfer, data, H323 to SIP signalling proxy, instant messaging, IVR, PC2Phone and Phone2PC gateway, SCCP — GTT routing between networks, Secure Unified Communications, SIP registrar-proxy, SIP SBC (session border controller), USSD, voicemail, VoIP, VoIP to PSTN gateway, conference server (max 200 voice channels per conference), call centre server, prepaid and postpaid cards | Deployed on home servers and large networks with millions of users | 5.5; May 2015; 4 years ago |
The following table is an overview of those VoIP clients which provide end-to-end encryption.
Client name | Development status | Open source client | End-to-end authentication[a] | Encryption protocols | Forward secrecy | Multiple encryption | Encrypted group calling | Proxy, Tor | |||
---|---|---|---|---|---|---|---|---|---|---|---|
ZRTP | ECDH | DTLS | SRTP | ||||||||
FaceTime | Active | No[72] | No[72] | ? | ? | ? | ? | Yes[72] | ? | No | No |
Google Duo | Active | No | No | ? | ? | ? | ? | ? | ? | No | No |
Jitsi[b] | Active | Yes | Yes[72] | Yes | No | Yes | Yes | Yes[72] | Yes | Yes | ? |
Line[73][74] | Active | No | ? | ? | ? | ? | ? | ? | ? | No[73] | No |
Linphone[b] | Active | Yes | Yes | Yes | ? | Yes | Yes | Yes | Yes | ? | No |
PGPfone | Abandonware | Viewable source[75] | Yes | ? | ? | ? | ? | ? | ? | ? | ? |
Signal | Active | Yes | Yes[72] | No | Yes | No | Yes | Yes[72] | Yes | No | Depends[76] |
Silent Phone | Active | Viewable source[77] | Yes[72] | Yes | ? | ? | Yes | Yes[72] | Yes | Yes | No |
Telegram | Active | Yes | Yes[78] | No | No | No | No | Yes | ? | No | No |
Threema | Active | Partially[c][79] | Yes[80] | No | Yes[80] | Yes[80] | Yes[80] | Partially[d][80] | Yes | No | No |
Viber[44] | Active | No | ? | ? | ? | ? | ? | ? | ? | ? | No |
Active | Partially[c][81] | Yes[81] | No | Yes[81] | No | Yes[81] | Yes[81] | Yes[81] | No | No | |
Wire | Active | Yes | Yes[82] | No | ? | Yes[49] | Yes[49] | ? | Yes | Yes | No |
Zfone | Abandonware | Viewable source[83] | Yes | Yes | Optional[84] | ? | Yes[85] | Yes | Yes | ? | ? |
The following table is an overview of those VoIP clients which provide client-to-server encryption.
Client name | Encryption protocols |
---|---|
Google Hangouts[72] | SRTP[86] |
Skype[72] | A custom protocol |
If you don’t have Viber on your phone and try to install the service on your PC, the app will redirect you to its website and ask you to install Viber on your phone first.