Last week, we announced something exciting coming to Gruveo, and the time has come to lift the curtain a little.
Next Monday, we will be launching direct Gruveo codes - a new way to use Gruveo for easy video and voice calling that is bound to make phone numbers a thing of the past.
What are the direct codes?
Unlike a traditional one-time code that anyone can use to establish a Gruveo call, a direct code - for example, @john - belongs only to you. Whenever somebody enters "@john" on Gruveo or follows your Gruveo link (www.gruveo.com/@john), you receive that as an incoming call which you can answer in the Gruveo app or on the Gruveo website.
Think of a direct Gruveo code as a supercharged toll-free phone number that your clients, customers or simply friends can use to instantly get connected to you without installs or accounts. Better yet, direct codes will be free to get and use, no strings attached.
Registration is opening on June 27th. Direct codes will be available on the first-come, first-served basis, so make sure to grab the perfect one for yourself or your business quickly! We are not holding anything back so even single-letter direct codes like @x will be available on day 1.
Learn more about direct Gruveo codes over here:[button_1 text="Direct%20Codes%20Explained%20›" text_size="20" text_color="#ffffff" text_bold="N" text_letter_spacing="0" subtext_panel="N" text_shadow_panel="Y" text_shadow_vertical="1" text_shadow_horizontal="0" text_shadow_color="#000000" text_shadow_blur="0" styling_width="25" styling_height="15" styling_border_color="#dd7103" styling_border_size="1" styling_border_radius="15" styling_border_opacity="100" styling_shine="Y" styling_gradient_start_color="#ffbe6b" styling_gradient_end_color="#dd7103" drop_shadow_panel="Y" drop_shadow_vertical="1" drop_shadow_horizontal="0" drop_shadow_blur="1" drop_shadow_spread="0" drop_shadow_color="#000000" drop_shadow_opacity="50" inset_shadow_panel="Y" inset_shadow_vertical="0" inset_shadow_horizontal="0" inset_shadow_blur="0" inset_shadow_spread="1" inset_shadow_color="#fece94" inset_shadow_opacity="50" align="center" href="https://about.gruveo.com/direct/"/]
WebRTC (Web Real-Time Communications) is an open source technology which allows two browsers to connect natively for peer-to-peer video and audio calls. Calling is possible without needing to download any additional software or install plugins...straight from your browser!
Where did WebRTC come from? In May 2011 an open source project was released by Google. With massive help from the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF) there was ongoing work to standardize protocols and improve browser APIs. Progress is still being made today.
So how does it work? WebRTC works peer-to-peer, meaning that it isn’t necessary for the signal to be relayed via a server. Call quality is often far superior to some previous solutions, where servers can be positioned in far-flung locations, leading to lag. WebRTC providers package this technology into easy-to-use systems which intelligently determine how to best stream (and relay, if necessary) the media from one user to the other, optimizing call quality and limiting latency.
There are several key benefits of using WebRTC over previous solutions for video conferencing:
A rising concern in business, a high level of security is often top of the list when IT departments are considering new software. In WebRTC it is impossible to have an unencrypted call. Even if a TURN server is required to relay the media stream, calls still remain encrypted end-to-end.
If an even higher level of control is required it is also possible to have a system using WebRTC installed directly on the company servers, operating within your own firewall. This enables both security and ease of use within a company and is ideal if you operate from more than one site.
As discussed before, you will probably find the call quality with a WebRTC-based platform superior, especially if you are calling a location not very geographically far away. This is because peer-to-peer technology operates directly from your computer or phone to the person you are calling. If there are any problems relaying the call, that’s when the TURN servers spring into action. Due to providers often having numerous servers in different locations around the world, call quality is generally well maintained.
If you have ever used a WebRTC-based app, one thing you may notice is how surprisingly easy it is to use. You don’t need to install anything...you can use it straight out the box. And it can be as simple as pressing the call button and sending a link or code over to the other party for almost instant video conferencing.
This ease of use makes WebRTC perfect for the workplace, general calls with technophobic friends and family, or integrating into a website for easy use for customers or clients.
Due to the benefits and ease of use, there are many situations where a WebRTC-based app is the perfect solution for a company:
So is WebRTC the future of video conferencing? With all the benefits of this technology it certainly seems that way. It is predicted that it will be available on 6 billion devices by the end of 2019, so chances are you are going to be hearing a lot about WebRTC over the next few years. With the variety of ways this technology can be utilized it is quite certain that video conferencing will be used even more extensively in companies all over the world.
Gruveo Business uses WebRTC technology to deliver the ideal solution for video conferencing. Want to find out more? Visit our Business site here.
You get a new app or system for the company, you go through the installation and user training. It’s ready to be used by everyone.
And you know – you just KNOW with absolute certainty, that this new tech is going to unleash a flood of daily support ticket requests. They will range from the most basic questions about how to turn it on to more complex ones.
That is exactly the case with video conferencing.
Today, video conferencing is a fact of business life. It is a must-have strategic tool that is quickly finding its way into every conference room and office.
Of course that means more support requests. Many more.
But it doesn’t have to be that way.
The following is a list of some of the most common problems we hear about and what you can do about them in your organization.
The list of potential glitches is long. It includes connectivity issues, setup problems, and audio and visual breakdowns.
Additionally, they should schedule an extra 5 minutes into the beginning of their live session to work out any problems they may experience once the session begins. That way, if glitches do arise and time is spent fixing them, the time won’t feel wasted.
Too many video conferencing solutions are counterintuitive and deliver aggravatingly complicated user experiences.
It is not uncommon for many video conferencing users to not even know how to start a video session. And when they do, they often become overwhelmed by all of the (unnecessary secondary) features.
Additionally, the average user is typically uninformed about important “under-the-hood” technology issues, including bandwidth and CPU, which directly impact session quality.
But one-time is not enough.
The training needs to be ongoing. Not everyone uses video conferencing all of the time. So if there is a lag between the time someone is trained and the time they actually use the system, he or she will likely forget how to use it and may then avoid using it altogether (leading to another common problem – adoptability).
It is helpful to have ongoing training in the form of webinars, videos, and an easy-to-follow 1-page cheat sheet, which should all be saved in one shared location.
It is a session breaker.
Unbeknownst to many, a prime culprit of this is their computer’s system resources such as CPU, RAM and bandwidth. Your average employee has no idea what that even is.
But if the system resources begin to get consumed – say by a bloated antivirus suite – a smooth video session can suddenly break down and turn into a ni...g..h…...t..ma…..re.
Those are just 3 common problems that can end up building up in your support ticket queue.
As you can see, primary antidote to these problems is education. Provide your users with lots of video tutorials, recorded webinars, FAQs, and cheat sheets all stored in a single location such as your company’s Google Drive.
We at Gruveo take our users’ privacy and security very seriously. In this blog post, we’d like to share some details on the technology behind Gruveo and the security and privacy measures we have in place.
Gruveo uses WebRTC for all video and voice calls made using its platform. WebRTC is a free, open technology that enables web browsers with Real-Time Communications (RTC) capabilities.
WebRTC is often described by the industry professionals as the most secure VoIP solution out there.
WebRTC specification requires that all transferred data – audio, video and custom application payloads – must be encrypted end to end while in transit. This is achieved by employing the following protocols:
DTLS is a privacy protocol that is very similar to TLS (SSL), but with a minimal number of modifications to make it compatible with the UDP transport used by WebRTC. DTLS enables a secure data channel between peers that cannot be tampered with. No eavesdropping or message forgery can occur on a DTLS encrypted connection.
SRTP is a secure variant of the standardized format for delivery of real-time data, such as audio and video over IP networks. SRTP media cannot be decrypted by a third party thus ensuring that IP communications across the Internet remain private. In other words, SRTP ensures that WebRTC voice and video traffic will not be heard or seen by unauthorized parties.
Finally, WebRTC is a peer-to-peer technology where calls are established directly between the peers’ devices for lower latency and added security. In some situations, a peer-to-peer call cannot be established and the call data has to travel through the Gruveo’s servers. However, DTLS and SRTP ensure that the call contents cannot be decrypted on the server even in such a scenario.
All text messages on Gruveo are relayed via Gruveo’s secure servers. The messages are relayed to and from client endpoints in encrypted form using TLS (SSL) as part of the WebSocket Secure (WSS) protocol.
The Gruveo website is only accessible via the secure HTTPS protocol.
Endpoint security is out of Gruveo’s control. For example, we cannot detect or prevent a virus running on a client machine from recording the user’s communications, on Gruveo or otherwise.
All Gruveo users are encouraged to choose longer, non-trivial codes for connecting to ensure against a random third party joining under the same code before the intended counterpart does.
Once a call between two parties has been established on Gruveo, no one else can connect to it, even if they enter the same code. Anyone connecting under the same code while you are talking will get a "busy code" message.
We hope that this has been helpful in understanding how Gruveo protects your privacy and security. If you have more questions, please don't hesitate to contact us right away.
Update August 15, 2014: The below post applies to an earlier version of Gruveo that used Flash. Click here for a detailed review of the current version's security.
2013 has brought revelations about the massive scale of US surveillance on Internet communications. People all over the world learned that most of their Internet activities are routinely logged, recorded and analyzed. In light of these revelations, we at Gruveo feel that it’s our duty to explain to our users how their calls are protected and what mechanisms we use to ensure the security of their Gruveo communications.
The first thing to note is that wherever possible, all Gruveo calls are established using the so-called peer-to-peer (P2P) technology where data flows directly between the users’ computers. Almost by definition, the absence of a middle point relaying your calls means that it’s harder for a third party to intercept them.
Whether a call is established using P2P is determined by the firewall configuration of the particular pair of users. If a call cannot be established via P2P, it is relayed using our secure servers. Gruveo is quite good in “piercing” firewalls to establish P2P calls, however: 63.1% of the past month’s calls were established over peer-to-peer.
The decentralization brought by peer-to-peer is just part of the story. No matter if your call is established via P2P or not, it is encrypted end to end as part of the RTMFP protocol (P2P), or RTMPS/RTMPTS (non-P2P). In RTMFP, all network traffic is encrypted using 128-bit cipher. RTMPS/RTMPTS rely on the industry-standard SSL standard for traffic encryption.
Last but not least, given how Gruveo connects users, isn’t it possible that someone else can connect to your call just by accidentally entering the same number while you’re talking to somebody? The answer is no – once a Gruveo call is established, it is “sealed” and anyone entering the same number will create a new session just as if your call didn’t exist.