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Kranky Geek WebRTC event summary 2022
Kranky Geek 2022 follows our tradition of great curated content on WebRTC that is both timely and timeless. Here’s what we had this year.
Kranky Geek is the main event focusing on WebRTC. I’ve been doing it with Chris Koehncke and Chad Hart for many years now, with the help and assistance of Google along with various sponsors each time.
Like many, we’ve switched to an all virtual event since the pandemic started, and decided at least for this year to continue in the same format. This turned out well, since I had to go on a business trip to Ireland at the date of the event, and virtual meant I was still able to both host and speak at the event.
Kranky Geek is quite a grueling experience for the hosts. We curate the sessions, at times approaching those we want to speak, at other times telling the speakers what topics we think will fit best. We go over the draft slide decks and comment on them. Doing dry runs on the week of the event with all speakers to make sure the session is top notch.
You won’t find much commercial content in a Kranky Geek event. What you will find is lots of best practices and suggestions based on the experience and the path taken by our great speakers.
To this year’s summary, Philipp Hancke did the commentary about the sessions themselves. If you are a WebRTC Insights subscriber, and would like to discuss the content and how it fits in your company, feel free to reach out to me to schedule a meeting.
If you are looking for the whole playlist, you can find it here. The videos have been embedded below to make it easier for you to watch.
Roundtable: The state of Open Source in WebRTC- Jitsi, Janus, mediasoup and Pion
- Watch if you are using any of these projects
Background blurring and light adjustment using MediaPipe.
- ML powered background blur and light adjustment using MediaPipe
- Performance in the browser is a challenge, as is model size
- A lot of data and practical examples. Not an introduction to MediaPipe though
- Watch after you built a background blur pipeline yourself and want to learn how to improve it
Krisp SDK on the Web: noise suppression.
- ML powered audio improvements on the Web
- 128 samples per frame suggest using WebAudio / Audio Worklets. Note that there’s a Chrome “L16 hack” which can deliver 10ms frames of “raw” audio
- Watch when you consider building an audio processing pipeline yourself as well as after your first attempt
Where I speak and Philipp comments (I am kinda subjective on this session).
- WebRTC’s getStats API is tremendously powerful but making sense of the numbers is quite a challenge. Turning those numbers into something actionable is quite tricky
- There are a lot of things that can go wrong and are actionable. The tooling testRTC has is quite valuable answering the common support questions. Troubleshooting is important but may depend on how much time you can spend on debugging a customer’s environment
- Watch when you are stuck trying to troubleshoot a problem with just WebRTC statistics and need to take a look beyond them
- A billion minutes every day. That sounds impressive. There is a but here however. In 2018 the number Google told was 2.5 billion per week or around 400 million per day (on weekdays and half on weekends?). That means a 2.5x growth in four years. With a pandemic in between. Meta said Whatsapp is doing 15 billion minutes per day… WebRTC in the browser remains small. One wonders what happened to their 100x usage (which was received minutes). We’re past peak usage
- Some Insights into their roadmap. We have been tracking most of this in WebRTC Insights over the past year so no big surprises if you are a subscriber
- Elad Alon provides a good overview of the improvements he did to screen sharing such as preferring tab sharing in the getDisplayMedia picker
- Markus Handell talks about a lot of things like Metronome which, as a reader of WebRTC Insights may mean something to you. Great slide that shows the pipeline and where the improvements to individual components affect it
- Harald Alvestrand gives a great summary of what new APIs are coming to WebRTC in general. Control about ICE candidates seems super interesting but we have not yet figured out what the field trials actually enabling
Recording and compositing video sessions.
- Great overview of the considerations you need to make when doing WebRTC recording
- The goal is to get a view like in the browser but you cannot record in the browser, not even on the server
- Watch if you are looking to build a recording feature and are interested in the requirements
- A great introduction to why streaming wants to have “standardized signaling”
- WebRTC was built with JavaScript and the flexibility in mind but native applications need a bit more standardization. And let us not talk about hardware encoders
- In many ways, WebRTC breaks the model of vendor’s ecosystem where media servers and device manufacturers of the past had to interoperate by having a single vendor take care of it all. For live streaming and broadcasting, this interoperability model is still very important
- Watch if you are interested in streaming use-cases
FlexFEC and video.
- NVIDIA uses video forward error correction for GeforceNOW, their game streaming service. The requirements for such a service are quite different from WebRTCs “talking heads”
- Video forward error correction is a surprisingly obscure topic in WebRTC
- Watch if you are interested in learning how FEC works and how to measure the improvements
- Audio remains the most important thing in conferencing – you need to understand what the other side is saying. This talk explains the history of Opus and how Lyra and Satin fit into the picture as well as how forward error correction and redundancy work
- The famous “Opus comparison” picture is quite problematic when looked at in detail
- Watch if you are interested in going beyond the usual Opus forward error correction mechanism available in WebRTC by default
It should be noted that without our sponsors, doing the Kranky Geek event would be impossible. When we set out to run these events, we had this in mind:
- Content should be free for all
- Participating in the event should be free or have a token cost associated with it
- Content must be top notch. Timeless. With little to no sales pitches
This requires sponsors to help with funding it. Each year we search for sponsors and end up with a few that are willing and happy to participate in this project of ours.
This year?
- Google, and especially the team working tirelessly on WebRTC
- Daily, who operates a video API (CPaaS) platform with a strong lowcode/nocode focus
- Krisp, with their AI solution to handle background voices, noises and echo
- Spearline, and their testRTC products for WebRTC testing and monitoring
Check them out
A Kranky Geek 2023?When we’re planning and preparing for the event, it feels like this is going to be our last event. It isn’t easy, and none of us in the Kranky Geek team are event planners by profession. The question arises after each such event – will we be doing another one?
Once this event was over, we started working on wrapping the event. Part of it was editing the content and uploading it to YouTube (which takes time).
Will there be another Kranky Geek event next year? Maybe
Will it be in person or virtual? Maybe
Until then, go check out our growing library of great WebRTC content: https://www.youtube.com/krankygeek
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WebRTC: Privacy or Privacy? Which one shall it be?
WebRTC comes with mandatory encryption, which enables privacy, but which type of privacy are you really looking for?
DALL-E: a broken lock on a chestIn the past, all the great stuff started in the enterprise and then trickled down to consumers. Now it is the other way around – first features come to consumers and from there find their way to enterprises.
Privacy is no different, but in enterprises it needs to be defined quite differently, making it a totally different kind of a feature.
This is where privacy vs privacy comes to play.
Table of contents- Privacy: The consumer version
- The enterprise version of privacy
- WebRTC and privacy
- Who cares?
- CPaaS, Video API and… privacy
- What next?
As a user, what do you mean when you say privacy?
That the data you generate is yours. Be it sensor related data (think GPS or heart rate). The conversations you have with people are not accessible to anyone else. The same for the photos you take.
Practically, you want no one other than you and those you explicitly share data with to have any access to that data. And that includes the services you use to generate and share that data.
Sending messages over Whatsapp or any other social media service? You probably want these messages to be encrypted on the go, so no one can sniff the network and read your messages. You also don’t want Whatsapp’s employees reading what you wrote.
Essentially, what you are looking for is E2EE – End-to-End Encryption. This means that any intermediary along the route of your communications, including the communication provider himself who is facilitating the session, won’t have the ability to read the content. Simply because it is encrypted using some encryption key that is known only to those on the session.
The enterprise version of privacyLife for a consumer is simple. At least when compared to an enterprise.
In the enterprise you want this privacy thingy, but somehow you also want governance and the creation of some corporate knowledge base.
When a meeting takes place. Should only the people in the meeting have access? Think about it. Should the people involved in that aspect of the business have access?
Let’s say we’re on a sales call with a customer. And then the sales rep on that call leaves and gets replaced with another one. Should the new sales rep have access to that call that took place and the decisions made in it?
Today, our CRM systems can connect directly to the corporate email and siphon any emails sent or received with certain customers into their account for recording and safekeeping. So we stay in sync with all conversations with that customer.
We may need to store certain conversations due to regulatory reasons. Or we might just want to transcribe them for later search – that internal company knowledge base repository.
There are also times when we’d like to use these conversations we’re having to improve performance. Similar to what Gong does to sales teams.
BUT
We don’t want others to have access to these meetings. In some cases, we don’t want the theoretical ability of the provider of the service to access these conversations – think of a Microsoft Teams session, Google Meet or a Zoom call that gets listened to by the employees of these companies.
Privacy in an enterprise looks different than for consumers. It is more granular and more structured, with different rules and permissions at different levels and layers.
WebRTC and privacyPrivacy is king in WebRTC, with a few caveats:
- Only if you let it
- Assuming you don’t screw it up
- When it is of interest to you
Why these caveats?
- Because WebRTC is just a building block – the actual solution is of your making. Which means you can screw it up by architecting or implementing it wrong
- It also means that you want to have privacy as part of your service
And why is privacy king in WebRTC? Because security is ingrained in WebRTC, which means you can use it to provide privacy conscious services.
Lets go over what privacy in WebRTC actually means:
WebRTC mandatory encryption (and security)In WebRTC, all media is encrypted. You can’t decide to send media “in the clear”. And then the signaling itself is also encouraged to be encrypted, and for all intent and purpose – it is encrypted as well.
This means that if you send audio or video via WebRTC from one user to another or from one user to a media server – then that media is encrypted and can be played only by the recipient.
Someone looking at the bitstream “over the line” won’t be able to play it back or intervene with the content.
Note here that a media server terminates the conversation here and is privy to what is being sent – it has access to the encryption keys. TURN servers don’t have such access.
This mechanism of encryption isn’t optional – it is just there.
E2EE in WebRTCIf we increase the scope to group conversations, then we need E2EE – End-to-End Encryption.
This can be achieved on top of WebRTC using a mechanism known as insertable streams, which ends up as double encryption – one between the sender and the media server. And one between the sender and the receivers on the other end. That second layer of encryption is part of the application. WebRTC doesn’t mandate it or even encourage it – it just enables you to implement it.
Deniability vs governance of communications in WebRTCHere’s where things can get tricky with WebRTC – it can be used to cater for both ends of the equation.
You can use WebRTC to obtain deniability.
WebRTC has a data channel that runs peer to peer. Using signaling servers to open up such connections to create a loose mesh network of peers means you can send private, encrypted messages from one user to another on that network without having any easy way to trace the communications – let alone to trace its metadata. That’s on the extreme scale of what can be achieved with WebRTC – a TOR/bittorrent-like network.
With the same methodology, I can get two users or even small groups to communicate directly, so that their media travels between them and them alone. Or I can employ E2EE on media servers and get privacy of the content of the communications from the infrastructure used to facilitate it.
You can use WebRTC to handle governance.
On the other side of the equation, you can use WebRTC and force all communications to go through media servers. Media servers which can then enforce policy, record media and provide governance. For some industries and verticals – that’s a mandatory requirement.
And you get these capabilities while keeping the communication encrypted over the internet.
Who cares?With privacy that’s the biggest question. Who cares?
No one and everyone at the same time.
If you ask a person if he wants privacy the immediate answer is – yes!
And yet… Twitter still doesn’t offer E2EE on DM messages. And people use it.
Whatsapp added E2EE in 2016, when it already had a billion monthly active users. It added E2EE backups in 2021. It seems people wanted it, but not in such high demand to switch to a more secure and private messaging system.
Here’s a screenshot from my own Whatsapp in one of the groups I have:
That weird message is an indication that a friend of mine has changed his security code. This usually means he re-installed Whatsapp or switched a phone I presume. I ignore these messages altogether, and I am assuming most people ignore these messages.
In the same way, companies want and look and strive for privacy and want the services they use to be private. But most of them want it up to a point.
Does that mean privacy isn’t needed? No.
Does it mean we shouldn’t strive for privacy? No.
It just means that people value other things just as much or even more.
CPaaS, Video API and… privacyWhen it comes to video APIs and CPaaS platform, it feels that privacy is somewhat lagging behind.
Messaging platforms today mostly offer E2EE. UCaaS are and have been introducing E2EE to their chat services and video calls. Some are offering integration with third party KMS (Key Management Systems) so they don’t have access to the decryption keys to begin with.
CCaaS relies heavily on the telephony network, where, well, what privacy exactly? And they also like to record calls for “quality and training purposes” – which translates to using machine learning and providing governance.
Video CPaaS is somewhere in-between these days – it offers encryption on sessions because it uses WebRTC, which is encrypted by default. But anything going through the media server can usually be accessed by the Video APIs vendor itself. Very few have gone ahead and added E2EE capabilities as part of their solution.
The reasons for that? It is hard to offer E2EE, but it is even harder to offer it in a generic manner to fit multiple use cases. And on top of that, customers don’t necessarily care or will be willing to pay for it, while they will be willing to pay for features such as recording.
What next?Here’s the thing:
Everybody talks about privacy but nobody does anything about it
In the consumer space, we are moving to an E2EE world.
The enterprise space is glacially pacing towards that same goal.
Parallel to that though, machine learning and cloud media processing are shifting the balance back towards less privacy – at least less privacy from the vendor hosting the service.
Which is more important to the buyers of services? Privacy or governance? Deniability or machine learning?
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Revealing mediasoup’s core ingredients: Q&A with Iñaki Baz Castillo
I interviewed mediasoup’s co-founder, Iñaki Baz Castillo, about how the project got started, what makes it different, their recent Rust support, and how he maintains a developer community there despite the project’s relative unapproachability. mediasoup was one of the second-generation Selective Forwarding Units (SFUs). This second generation emerged to incorporate different approaches or address different use cases a few years after the first generation of SFUs came to market. mediasoup was and is different. It is node.js-based, built as a library to be part of a serve app, and incorporated the Object-oriented approaches used by ORTC – the alternative spec to WebRTC at the time. Today, mediasoup is a popular SFU choice among skilled WebRTC developers. mediasoup’s low-level native means this skill is required.
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Two years of WebRTC Insights
It is time to stop for a second and review what we’ve accomplished here with our WebRTC Insights in the past two years.
There are a few pet projects that I am doing with partners, and one of the prime partners in crime for me is Philipp Hancke. We’ve launched our successful WebRTC codelab and are now in the process of finalizing our second course together – Low-level WebRTC protocols.
Two years ago, we decided to start a service – WebRTC Insights – where we send out an email every two weeks about everything and anything that WebRTC developers need to be aware of. This includes bug reports, upcoming features, Chrome experiments, security issues and market trends.
All of this with the intent of empowering you and letting you focus on what is really important – your application. We take care of giving you the information you need quicker and in a form that is already processed.
Now, two years in, it is safe to say that this is a VERY useful tool for our subscribers.
“WebRTC insights might be the most important email you read every fortnight as a RTC / video engineer. It’s hard to keep tabs on what Google et al are doing with WebRTC while working on your product and the WebRTC Insights provides very specific and actionable items that help tremendously. We have been ahead countless times because of it. If you are serious about WebRTC you should definitely subscribe 100% worth it.”
— Saúl Ibarra Corretgé, Principal Software Engineer @ 8×8 (Jitsi) How do we keep track of all the WebRTC changes?Keeping track of all the changes in WebRTC is a pretty daunting task. Tsahi started WebRTC Weekly almost nine years ago and it has been the source of high-level information ever since. Philipp has closely worked with WebRTC at a more technical level for a decade too. We both had our routines for keeping notes and transforming them into something informative for our audience but joining forces (which we never expected after having strong arguments about whether XMPP was a great signaling protocol in the early days!) has yielded a surprising amount of synergy effects.
We start doing Insights with a template. Whenever we find something that we think is interesting we add a link and maybe a very brief comment to that template . Usually we chat about those too (as we have done for…. almost a decade now). Then we move on because both of us have day jobs that keep us busy.
Every two weeks we spend a couple of hours turning the “brain dump” into something that our audience understands. Philipp focuses on the technical bits while Tsahi focuses on the market. Then we review each other’s section, improve and exchange thoughts.
We did this before Insights already but putting a structure and a biweekly cadence to it has “professionalized” it. While it remains a side project for us, we now have the process in place.
WebRTC Insights by the numbersWe’re not new to this, as this is our second year, we might as well also compare the numbers today with those we’ve had on year one of WebRTC Insights:
26 Insights issued this year with 447 issues & bugs, 151 PSAs, 11 security vulnerabilities, 146 market insights all totalling 239 pages. We’ve grown on all metrics besides security vulnerabilities.
WebRTC is still ever changing, but at least there are less security threats in it
Activity on libWebRTC has cooled down a bit in the last two years when it comes to the number of commits and people working on it:
After more than a decade that is a sign of maturity, the easy changes have already been done and all that is left is optimizations. The numbers we see for Insights roughly correlate with the amount of energy Google puts into the project. We are just glad we did not start it during the “hot phase” of 2016-2019.
Let’s dive into the categories, along with a few new initiatives we’ve taken this year as part of our WebRTC Insights service.
BugsAmong the really useful feedback we have received was the suggestion to add a “component” or area the issue is in. This is useful for larger teams where one person may be digesting the biweekly email and route this to a subteam with a particular focus such as audio, video or networking.
The other improvement is a visual hint whether a particular item is a bug, a regression, a feature or just something that is generally good to know:
In addition to that we classify it as “read, plan or act”. Of course we hope our subscribers read all the issues but some are more important than others.
PSAs & resources worth readingPublic service announcements or PSA are the main method Google’s WebRTC team uses to announce important changes on the discuss-webrtc mailing list. We track them and give some context why they are important or whether they are safe to ignore (which can happen for API changes where a PSA may be required by the release process.
We also look at important W3C changes in this section as well as other content that is too technical for the “market watch” section.
Experiments in WebRTCChrome’s field trials for WebRTC are a good indicator of what large changes are rolling out which either carry some risk of subtle breaks or need A/B experimentation. Sometimes, those trials may explain behavior that only reproduces on some machines but not on others. We track the information from the chrome://version page over time which gives us a pretty good picture on what is going on:
In this example we saw the AV1 decoder switch from libaom to libdav1d over the course of several weeks.
WebRTC security alertsThis year we continued keeping track of WebRTC related CVEs in Chrome (totaling 11 new ones in the past year). For each one, we determine whether they only affect Chromium or when they affect native WebRTC and need to be cherry-picked to your own fork of libwebrtc when you use it that way.
To make it easier to track, we now keep a separate Security Tracker file that gets updated with new issues as they are found. This makes it easier to glance at all the security issues we’ve collected.
On top of that, when there’s a popular open source component that has its own security issues published, we tend to also indicate these, though not add them to the Security Tracker, so they aren’t even counted in our statistics.
WebRTC market guidanceInformation overload. That’s what all of us face these days with so much material that is out there on the Internet. On our end, we read a lot and try to make sense of it.
Part of that is taking what feels relevant to WebRTC and sharing it with our WebRTC Insights subscribers. It includes the reference to the article, along with our thoughts about it.
For product managers, this is their bread and butter in gleaning the bits and pieces of information they need to make educated decisions about roadmap and priorities.
For developers, this brings a bit more context than they are used for in their daily work – and is often outside of their immediate work and expertise.
Our purpose? Enrich your world about WebRTC and express some of the power plays and the shifts in the market that are taking place. So you know them well ahead of them happening in force.
Covering important eventsWe really enjoyed Meta’s RTC@scale event. In terms of quality and technical depth it set a bar for the upcoming KrankyGeek event which had been the gold standard so far.
However, the technical depth of the event was too intense for it to be digested in real-time. This meant Philipp sat down on a rainy Saturday and started rewatching the videos while keeping notes. And ended up watching each session multiple times since there were so many great points that needed or even demanded a bit more explanation. This turned into a nine page summary of the event, annotated with the timestamps in the video.
We decided to make this summary public because, while we thought it provided a ton of valuable lessons to our subscribers. Meta made the content freely available and so should we. And hey, we keep referencing this every other week.
This may have been a one-off but we still genuinely enjoyed it so might repeat the exercise… on a rainy saturday!
WebRTC release notes interpretationWe started playing around with video release notes at the end of our first year, and quickly made it a part of the WebRTC Insights service.
Whenever Google publishes a release notes for WebRTC, we publish our own video with a quick analysis of the release notes (and the release itself) for our Insights clients.
We go over the release answering 4 main questions:
- Is the new release more about features or stability?
- What are the things developers should investigate in the new release?
- Which bugs and features in the new release should developers beware from?
- What can be disregarded and ignored in the release?
Our intent here, as with anything else, is to reduce the amount of work our clients have to do figuring out WebRTC details.
We are also making these release notes videos publicly available, 3-4 versions back, so you can derive value from them. You can find them on YouTube:
https://www.youtube.com/watch?v=DQt_OQT4ZAo&list=PL7fuFATIj-PUtMVTQKpW_odTCO0_CPfXV
Be sure to subscribe to receive them once they get published freely to everyone.
Join the WebRTC expertsWe are now headed into our third year of WebRTC Insights.
Our number of subscribers is growing. If you’ve got to this point, then the only question to ask is why aren’t you already subscribed to the WebTRC Insights if WebRTC interests you so much?
You can read more about the available plans for WebRTC Insights and if you have any questions – just contact Tsahi.
Oh – and you shouldn’t take only our word for how great WebRTC Insights – just see what our readers have to say about it:
“For any Service Provider or Apps who heavily relies on WebRTC, the WebRTC Insights offers great value. […] What I like most about the Insights is its bi-weekly cadence, which fits the rapid Chrome/WebRTC release cycle, and most of the mentions are actionable for us. With the recent Safari audio breakage, the Insights highlighted the problem timely and saved us a lot of troubleshooting effort.”
— Jim Fan, Engineering Director @ Dolby Laboratories“As a service company specialized in WebRTC I think WebRTC Insights is really useful. It keeps us up to date about what is coming next, giving good ideas for projects and research. Also, receiving periodic insights is always a good excuse to stop what I am doing and find some time to go over the latest WebRTC updates in more detail. It is much easier to do when you get all summarized in a single document than on your own just googling and going through an overwhelming list of webrtc news, updates and bugs.”
— Alberto Gonzalez Trastoy, CTO @ WebRTC.venturesHere’s the summary of the first year of Insights if you’re interested
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The lead actors in WebRTC are outside of your control
When developing with WebRTC, make sure you address the fact that many aspects are out of your control.
[In this list of short articles, I’ll be going over some WebRTC related quotes and try to explain them]
When you develop a WebRTC application, you need to take into consideration the sad truth that most of the things that are going to affect the media quality (and by extension the user experience) are out of your control.
To understand this, we first need to define who the lead actors are:
The main entities in WebRTC applications, taken from my presentation on testRTC Your applicationThis is probably the only piece you do control in a WebRTC application.
The code and logic you write in the application has immediate effect over the media quality and connectivity.
Deciding on how group calls are architected for example –
- Do you create a mesh network where everyone talks to everyone directly?
- Do you use a central mixer (MCU) to mix all media content to generate a single stream for each participant?
- Do you route the media using an SFU?
- What configuration limits do you impose on the streams on the get go?
- What kind of layout do you use to display the participants?
All these are going to greatly change the experience and it is all up to you to decide.
The browsersWeb browsers are out of your control.
I’ll repeat that for effect:
Web browsers are out of your control.
You can’t call Google asking them to delay their Chrome release by a week so you can solve a critical bug you saw cropping up in their upcoming release. It. doesn’t. work. this. way.
Browsers have their own release cadence, and it is brutal. In many cases, it is way faster than what you are going to be able to manage – a release every month.
The problem isn’t with this fast pace. It is with the fact that now, even after over 10 years since its announcement, WebRTC is still getting changed and improved quite frequently:
- Some of these changes are optimizations
- Others are bug fixes
- A few are behavioral changes
- Then there are the API changes to make the browser work closer to how the WebRTC specification states
All of these changes mean that your application might break when a new browser version goes out to your users. And as we said, you don’t control the roadmap or release schedule of the browser vendors.
The networkYou decide where to place your servers. But you don’t get to decide what networks your users will be on.
I often get into talks with vendors who explain to me the weird places where they find their end users:
- In an elevator
- Sitting in basements
- Driving a car on the highway
- At the beach
- In the library
- …
I am writing this article while sitting in the lobby of a dance studio on my laptop, tethered via WiFi to my smartphone’s cellular network (a long story). Users can be found in the most unexpected places and still want to get decent user experience.
WebRTC being so sensitive to the network connection (think latency, jitter, packet loss and bandwidth), these are things you’ll need to come to terms with.
In some cases, you can instruct your users to improve their connection. In others you can only guide them. In others still your best bet is to make do with what the user has.
Oh – and did I mention that the network’s conditions are… dynamic? They tend to change throughout the duration of the session the users are on, so whatever you decide to do needs to accommodate for such changes.
The user’s deviceIs your user running on a supercomputer? Or a 2010 smartphone? Do you think that’s going to make a difference in how they experience your WebRTC sessions?
WebRTC is a resource hog. It requires lots of CPU power to encode and decode media. Memory for the same purpose. It takes up bandwidth.
Your users don’t care about all that. They just want to have a decent experience. Which means you will need to accommodate for a vastly different range of devices. This leads to different application logic that gets selected based not only due to the network conditions, but also based on the performance of each and every user’s device – without sacrificing the experience for others.
Sounds simple? It is. Until you need to implement it.
How do you take back control of WebRTC?First step to gain control of your WebRTC application and its lead actors is by letting go.
Understand that you are not in control. And then embrace it and figure out how to make that into an advantage – after all – everyone is feeling these same pains.
Embracing them means for example:
- Testing on beta and dev releases of browsers, so that you’re more prepared for what’s to come
- Making sure you can upgrade your infrastructure and application at a moment’s notice with urgent patches
- Monitoring everything so you can understand user experience and behavior
- Place your servers closer to your users
- Optimize your application to work with different devices and networks within the same session
- Check for dynamic network changes and how that affects your service
- …
The WebRTC Developer training courses touch a lot of these issues while teaching you about WebRTC
My WebRTC Scaling eBooks Bundle can assist you in figuring out some of the tools available to you when dealing with networks and devices
This blog is chock full with resources and articles that deal with these things. You just need to search for it and read
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How Cloudflare Glares at WebRTC with WHIP and WHEP
WebRTC blackbox reverse engineering experts Gustavo and Fippo take a look at Cloudflare's new WebRTC implementation, how Cloudflare uses the new WebRTC-based streaming standards WHIP and WHEP, and the bold pronouncement that they can be a replacement to open source solutions.
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WebRTC turns services into features
Telephony and communications used to be services, but WebRTC has turned them into features inside other services.
[In this list of short articles, I’ll be going over some WebRTC related quotes and try to explain them]
Telephony and communication used to be services.
Need a phone system for your company? Go to your carrier, and they’ll set you up with a solution. Maybe install a PBX or even host one “in the cloud” for you.
The thing is, what you got was a full fledged service from a communication vendor. You had your own service, and your phone service. They were unlikely to be really connected to each other.
Then came along CPaaS vendors and communication APIs. You could purchase phone numbers and automate and route them anyway you wanted programmatically. In some ways, you could connect it to your own service logic, but only to a certain degree. If you had a call center agent who needed to answer a phone, you had to give him a physical phone (or install a softphone application for him), as well as the CRM application he interacted with in order to assist people calling in.
Two separate services.
Communications and Customer Relation Management (CRM).
WebRTC changes all that
Since WebRTC runs inside a browser, it lets you place the communication part right there in the browser, where all your other services live already.
It means that now that “telephony service” you had can be added as a feature inside your CRM.
But it gets better.
It doesn’t need to be a CRM you integrate it with. It can be a dating service. A gaming experience where you need to communicate with others during the game. A doctor visit at the virtual clinic. Remotely driving a car. Gambling online. The list goes on.
The main attraction isn’t the communication, but rather the service you are there to use. It so happens to need the ability to communicate using voice or video in real time, but that’s just a detail – a feature – no longer the service itself.
And that’s the real paradigm shift that WebRTC has brought with it.
My free WebRTC for Business People report is a great place to dive deeper into what WebRTC is and the changes it brings with it – the ecosystem around it and what companies are doing with it. Check it out.
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Post-Peak WebRTC Developer Trends: An Open Source Analysis
WebRTC had its peaks during the pandemic, but how is it doing now? Did all those new projects die, putting the community back at pre-pandemic “normal” levels or is WebRTC still going strong? I built and analyzed a dataset from over a million GitHub’s events since 2019 to help answer are there many new WebRTC-related repos, how many new users is WebRTC attracting, is the community coding as much as it used to, how are new API's like Insertable Streams and WebCodecs doing?
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WebRTC is the most secure VoIP protocol
WebRTC security and privacy are top of mind. You won’t find any other open standard VoIP protocol as secure as WebRTC.
[In this list of short articles, I’ll be going over some WebRTC related quotes and try to explain them]
Time for a quick security check…
Here are some concepts that are true when it comes to security, privacy and WebRTC:
- Security often requires sacrificing privacy
- Privacy often requires sacrificing security
- WebRTC is an attempt to balance the two, and let the application developers figure out which one their focus is going to be on – without sacrificing either security or privacy more than is needed in the process
But what does that exactly mean?
You remember that WebRTC is only a building block. Right? This means that it can’t offer full privacy or full security, since there’s an application developer on top, who can… well… screw things up.
If your developers don’t think about the security and privacy necessary, then your WebRTC application will look like this:
But if they do think about it (and they should, no matter what they are developing), then you should have security and privacy nailed down properly.
What WebRTC gives you when it comes to security and privacy?
- Encryption at transit
- Traffic is always encrypted between one WebRTC entity and another
- It is up to you to figure out how to maintain it if you need to – for example, using media servers likely means media is available in the clear on the media server
- Short development cycles
- WebRTC has a new version released every month – because that’s the release cadence of Chrome
- It means the client code on the browser can be refreshed and updated frequently, which makes patching up security issues easier on that front
- You will need to figure out how your own release cadence for your native clients and your server infrastructure, especially when it comes to security patches
- Open implementation
- This means people can scrutinize the actual protocol and its implementation
- Over time, this leads to more secure solution, as more eyeballs can review what’s going on
- You can learn more about open vs closed security here
- Shoulder of giants
- Google Chrome, Apple Safari, Microsoft Edge, Mozilla Firefox
- Together they power most of our browsable internet
- They do it at scale, and securely (for the most part)
- All of them integrated WebRTC into their browsers, and they adhere to high security standards
- Would you rather trust a proprietary solution from an unknown/smaller third party instead?
- Modern
- Other VoIP standards are older
- As such, they were conceived and written before our modern era of cloud and smartphones
- This means they adhere to different threat surfaces than what is needed today
Need security?
WebRTC has the mechanisms available for you
- It is encrypted
- Requires signaling to be encrypted
- Enables end-to-end encryption via media servers by using Insertable Streams
Need privacy?
WhenRtc has the mechanisms available for you here as well
- It is encrypted
- Can run peer-to-peer, without any media servers touching the media itself
- You can use the data channel to “hide” data from passing through servers altogether (once a connection is established between the peers)
- Your decision on where you install and manage your infrastructure to add to the privacy you offer
Care about security? WebRTC is your best choice moving forward. But it won’t take the responsibility off your back.
Two pointers for you before you go:
Everything you need to know about WebRTC security
Zoom’s past security issues and why WebRTC is different
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Video API, CPaaS, programmability and WebRTC
A common solution for real time video apps is to rely on Video API, a niche in CPaaS, which makes use of WebRTC.
WebRTC has been around for enough time now to garner the creation of an ecosystem around it – both commercial and open source. I’ve recently covered the state of open source WebRTC solutions. It is time to look at the commercial solutions, and in them, to focus on the managed Video API – also known as CPaaS or “programmable video”.
The need for managed video API in WebRTCWebRTC isn’t simple. It is simpler than building the whole thing by yourself, but getting it deployed and building and managing all the backend side for it is a grind. It isn’t just the initial implementation and setup, but rather the ongoing updates and maintenance – WebRTC is still a living thing that requires care and attention.
Some vendors go for open source solutions. Others will aim for commercial infrastructure that they host on their own. Most would simply rely on the cloud, using a video API.
At the end of the day, the concept is rather simple – a third party CPaaS / video API vendor puts up the infrastructure, maintains it and scales it. Adds a public API on top. Sprinkles helpful documentation. And gets developers to pay for the use of this infrastructure.
It is a win-win for everyone:
- The video API vendor has paying customers, while focusing on delivering high real time video communication solution
- Developers can use the video API to get faster time to market, while enjoying the video API vendor’s economies of scale and expertise. At the same time, they can focus on building their own application, where video is just one part of the whole experience
One thing to note is that there’s no specific definition or term to use here.
Some use video API, which I decided to use for this article.
Others will simply say CPaaS – Communication Platform as a Service, but refer to the video feature/product within that market. CPaaS does a lot more and usually focuses on voice and SMS.
There are those who use VPaaS – Video Platform as a Service, to try and explain that this is still CPaaS but for video. Or still UCaaS (Unified Communications as a Service) but for video. I never did relate to this one.
Others still use Programmable Video.
Then there’s RTC (Real Time Communication) and RTE (Real Time Experience). Trying to broaden the scope beyond the mere use of APIs.
I’ve used WebRTC API Vendor or WebRTC PaaS in the past. Today I am just trying to use CPaaS or video API. Mostly.
Video Call, Video Chat or Group Video?In the same manner that we have multiple names to describe video API vendors we have multiple phrases to describe what it is we are doing with real time video communications.
The usual names are video call, video chat, group video, video conferencing.
And then there’s also live streaming and broadcast – when a single or a small number of users broadcast their video in real time to a potentially very large audience.
Who are the video API vendors and what do they offer?Assuming you are looking for a video API vendor, who are the candidates we have in the market? Here are a few of them.
Twilio video APIYou can’t start any discussion about CPaaS without looking at Twilio. Twilio is the uncontested leader in CPaaS and communication APIs. It has grown up in that space and is expanding it beyond the initial developers and API focus it once had.
When it comes to video API, Twilio’s main offering is Twilio Programmable Video. It wasn’t the first to come to market, but it can’t be ignored simply because it comes from Twilio.
Sadly tough, the recent downsize at Twilio was accompanied with an email and a blog post shared by Jeff Lawson, CEO of Twilio:
As we’ve discussed frequently, we have four priorities for reaching profitability and leading in customer engagement: Investing in our platform reliability and trust, increasing the profitability of messaging, accelerating Segment adoption, and scaling the Flex customer base.
Twilio’s priorities are:
- Platform reliability
- Profitability of messaging
- Segment adoption
- Flex adoption
None of it really related to video API or the Twilio Programmable Video…
This doesn’t say that Twilio Programmable Video is good or bad. Just that it isn’t the main focus for Twilio.
Vonage Video APIThe Vonage Video API is another popular choice. It came to Vonage through its acquisition of TokBox from Telefonica. At the time, the TokBox API was one of the most widely known and used alternatives out there.
Today, the Vonage Video API is still going strong.
Vonage was acquired by Ericsson this year, which can be seen as either a good thing or a bad thing for the Vonage Video API.
On one hand, Ericsson doesn’t cater developers, the long tail or video calling use cases, so what do they have to contribute here? This is again just going to be a distraction for them.
On the other hand, Ericsson just acquired Vonage. They are unlikely to make big sweeping changes. As the world goes into recession, this may mean that the Vonage Video API internal resources will be left untouched a while longer compared to other vendors in this space.
The rest of the video API packThere are many other alternatives in the video API space.
In 2020 I shared my viewpoint as to the entrance of Amazon’s Chime SDK and Microsoft’s Azure Communication Services into this market of video API.
Other vendors include Agora, Daily, Dolby and many others.
Each with its own focus areas, strength and limitations.
What about Zoom video conferencing API? Is Zoom the exception to prove the rule?No list of video vendors will be complete without mentioning the elephant in the room – Zoom.
Zoom offers a set of APIs and integrations in various levels and uses:
- Zoom has an SDK to develop Zoom Apps – applications that live inside Zoom and interact with it
- Zoom Meeting SDK – letting you embed the Zoom experience inside your own application. Practically whitelabeling the Zoom interface
- Zoom Video SDK – a video API that is comparable and competitive with the rest of the vendors mentioned here
The biggest notable trend in the video API domain is the introduction of Prebuilt solutions.
I’ve been waiting for this to take shape for many years now, and it finally is happening.
Prebuilt are lowcode/nocode solutions that enable developers to write less code in order to embed the video API experience into their application. Here, the vendor offers more than an API layer, and instead of letting the developers using its platform figure out how to implement the UX/UI layer, it is given as a prebuilt component – usually with some level of configuration.
Most video API vendors today offer this in some form or shape or another – from official and unofficial reference applications, to iframe solutions, UIKits or application builders.
Daily, who has kindly sponsored my free ebook on nocode/locode in CPaaS is one such vendor. Their Prebuilt solution is quite comprehensive.
Is there a perfect video API?No.
Video communication is varied and flexible. It has many different scenarios and use cases. Today, there is no vendor that can cover all these use cases well. It means that there is no single vendor that offers a video API that can be automatically recommended for use.
The answer is more complex than that and boils down to “it depends”. It depends what it is you want to develop and what are your exact requirements and limitations.
There’s an (updated) report for thatTo understand which video API vendor offers the best fit to your needs, you can look at my Choosing a WebRTC API Platform report. It just got a fresh update.
25 vendors are covered, looking at them from various aspects. You’ll be learning:
- The various strategies developers tend to take in building their WebRTC applications
- What makes developers go for a video API solution, and what type
- Which KPIs you should be measuring in video API products
- What do each of the video API vendors have to offer you
This can greatly reduce the time it will take you to make your selection, as well as lower the risks of making the wrong decision.
Did I mention there’s a discount until the end of the month?
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With WebRTC, better stick as close as possible to the requirements, architecture and implementation of Google Meet
When developing with WebRTC, try to stick as close as possible to how Google Meet is designed and architected. That’s where love and attention is given to the source code.
[In this list of short articles, I’ll be going over some WebRTC related quotes and try to explain them]
Video is a resource hog. Some say that WebRTC is a great solution for 1:1 calls, but is lacking when it comes to group calling. To them I’d say that WebRTC is a technology and not a solution. In this case, it simply means that you need to invest some effort in getting group video calling to work well.
What does that mean exactly? That you need to think about bandwidth management first and foremost.
Why?
Let’s assume a 25 participants video call. And we’re modest – we just want each to encode his video at 500kbps – reasonable if we plan on having everyone at a mere VGA resolution (640×480 pixels).
Want to do the math together?
We end up with 12.5Mbps. That’s only for the video, without the overhead of headers or audio. Since we only need to receive media from 24 participants, we can “round” this down to 12Mbps.
I am sure you have a downlink higher than 12Mbps, but let me tell you a few things you might not be aware of:
- A downlink of 100Mbps doesn’t mean you can really get sustainable 12Mbps for a long period of time
- It also doesn’t mean you can get 12Mbps of incoming UDP traffic (and you prefer UDP since it is better for sending real-time media)
- Most likely, your device won’t be able to decode 12Mbps of video content at reasonable CPU use
- And if you have hardware acceleration for video decoding, it usually is limited to 3 or 4 media streams, so handling 24 such streams means software decoding – again running against the CPU processing limit
- The larger the group the more diverse the devices and network connections. So you’ll be having people joining on old devices and smartphones, or with poor network connections. For them, 12Mbps will be science fiction at best
- As a rule of thumb, I’d look at any service that uses over 3-4Mbps of downlink video traffic for video group calls as something that wasn’t properly optimized
You can get better at it, trying to figure out lower bitrates, limit how much you send and receive and do so individually per participant in the video group meeting. You can take into consideration the display layout, the dominant speaker and contributing participants, etc.
That’s exactly what 90% of your battle here is going to be – effectively managing bandwidth.
Going for a group video calling route? Be sure to save considerable time and resources for optimization work on bandwidth estimation and management. Oh – and you are going to need to do that continuously. Because WebRTC is a marathon not a sprint
Scaling WebRTC is no simple task. There are a lot of best practices, tips and tricks that you should be aware of. My WebRTC Scaling eBooks Bundle can assist you in figuring out what more you can do to improve the quality and stability of your group video calling service.
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CPO at Spearline and what it means to BlogGeek.me
I am now CPO (Chief Product Officer) at Spearline. This means that there are going to be some changes here at BlogGeek.me. Here’s what you can expect
Me, somewhere in Ireland, 3 weeks agoAlmost a year ago, testRTC, the company I co-founded, got acquired by Spearline. During that time, I got to know the great team there and the huge opportunity that Spearline has.
Since the above feels corny and a cliché to me as I write it, I’ll stop here.
To make a long story short:
- Spearline acquired testRTC (Spearline has its HQ in Ireland)
- Now they had 2 separate product lines: Voice Assure and testRTC
- As time went by, it was apparent that 2 is just a beginning
- And also that someone needs to manage product management as a whole
- Which is where I came in – they asked, and I said yes
- So now I am CPO at Spearline
First off, I am excited. Very.
It has been some time since I had a team to work with as their direct manager. It will also be the first time I get to manage product managers.
It also means that I am going to be investing a lot more of my time and attention at Spearline. Which is great, as I really love interacting with the people there already (I wouldn’t have accepted the role otherwise).
For my consulting business, it means that I will be shrinking it down considerably. I won’t be doing much consulting moving forward. It is somewhat sad, as I really loved helping people and hearing their stories and challenges. Hopefully, I will still get to do it in other ways.
What is going to stay, are all the initiatives that have taken place around BlogGeek.me over the years:
- My writing here on this blog will continue, though probably at a lower frequency
- The courses and reports will continue to be supported and updated. Me and Philipp Hancke are working to complete the new Low-level Protocols Course and we have plans for a few other courses after this one
- In the same token, WebRTC Insights is going to continue as a service
- And so will WebRTC Weekly and the Kranky Geek events
- From time to time, I’ll probably run an initiative or two here. Because I just can’t stop myself
All in all, it is time to continue and grow, and in a direction I have never expected I’ll find myself again.
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The WebRTC Developer Tools Landscape 2022 (+report)
An updated infographic of the WebRTC Developer Tools Landscape for 2022, along with my Choosing a WebRTC API Platform report.
This week I took the time to update my WebRTC Developer Tools Landscape. I do this every time I update my report, just to make sure it is all aligned and… up to date.
A few quick thoughts I had while doing this:
- Vendors come and go
- We see this all the time
- At the time of writing, I am aware of 2-3 additional changes that couldn’t fit to this update simply because of timing
- Testing & Monitoring is becoming more important
- There are more vendors there than they used to
- With my testRTC hat on, I can say this is a good thing
- Especially since we’re the best game in town
- CPaaS is crowded
- And becoming more so
- Is there room for everyone there?
- How will this market look like moving forward?
- Who should you be selecting for your next project?
- All these questions is what I am covering in the WebRTC API report
The WebRTC Developer Tools Landscape will never be complete. People always get pissed off at me when I publish it, not understanding why their company isn’t there. My answer to this is a simple one – because I don’t know what it is that you are doing.
They then get even angrier. What they should do at that point is ask themselves why I don’t know them enough. I have lived and breathed WebRTC since it was first announced. So if I don’t know their company and product, how do they expect others to learn about them?
I don’t think I am unique or special. Just that if you want to be in a landscape infographic that covers WebRTC, you might as well want to make sure people who deal with WebRTC and help others figure out what tools to use will know what it is that you’re doing.
What about that report?The report has been going strong for some 8 years now, with an update taking place every 8-12 months. It has been 12 months, so it definitely needed an update.
2 vendors were removed from the report and 3 new vendors added.
I’ve also decided to “upgrade” the term Embed/Embeddable/Embedded to Prebuilt. The reason behind it is the progress and popularity of these types of solutions in the video API space. Most CPaaS vendors today that offer a video API are also offering some form of higher level abstraction in the form of a ready made application – be it a full reference app, a UIKit, or a Prebuilt component.
The report will be published on 22 September. If you want to purchase it, there’s a 20% discount available at the moment – from now and until its publication.
Check out more about my Choosing a WebRTC API Platform report.
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Media compression is all about purposefully losing what people won’t be missing
With WebRTC, we focus on lossy media compression codecs. These won’t maintain all the data they compress, simply because we won’t notice it either.
[In this list of short articles, I’ll be going over some WebRTC related quotes and try to explain them]
The purpose of codecs – voice and video – is to compress and decompress the media that needs to be sent over the network. This was true before WebRTC and will stay true after WebRTC.
Generally speaking, there are two types of compression:
The two types of codecs- Lossless compression – these are codecs that whatever they see as input to the encoder will be generated in the other end of the decoder. Nothing will get lost along the way. Think of it as a .zip file – it stores files and requires a perfect match on both ends of the compression
- Lossy compression – these are codecs that don’t maintain an exact match from what goes into the encoder with what ends up after the decoder. These types of codecs are quite common with audio and video processing
Audio and video tend to hold a lot of data. And since we want to send it over the network, we’d rather not waste network resources. So what do these codecs do? They try to remove anything and everything that they can which our eyes and ears won’t notice much.
On a conceptual level, lossy compression has this virtual dial. You move the dial to decide how much you are willing to lose out of the data. The encoder will do its best to lose things you wouldn’t notice, but at some point, you’ll notice.
This flexibility in setting the compression level is also used to manage the bitrate. By estimating the bandwidth, the encoder can be instructed to turn the dial up and down the compression level to generate higher or lower compression to meet the requirements of the estimated available bandwidth.
Looking to learn more about video codecs? Go ahead and read my WebRTC video basics article
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The state of WebRTC open source projects
WebRTC open source is a mess. It needs to grow out of its youth and become serious business – or gain serious backing.
This article has been written along with Philipp Hancke. We cooperate on many things – WebRTC courses (new one coming up soon) and WebRTC Insights to name a few.
—
WebRTC is free. Every modern browser incorporates WebRTC today. And the base code that runs in these browsers is open sourced and under a permissive BSD license. In some ways, free and open source were mixed in a slightly toxic combination. One in which developers assume that everything WebRTC should be free.
The end result? The sorry state in which we find ourselves today, 11 years after the announcement of WebRTC. What we’re going to do in this article, is detail the state of the WebRTC open source ecosystem, and why we feel a change is necessary to ensure the healthy growth of WebRTC for years to come.
Table of contents- Your open source Cliffs Notes
- The WebRTC open source landscape
- WebRTC open source client libraries
- Open source TURN server(s)
- Open source signaling servers for WebRTC
- Open source SFUs and media servers in WebRTC
- Other, less popular open source alternatives for WebRTC
- Is it time for WebRTC open source to grow up?
We’ll start with the most important thing you need to know:
Open Source != Free
Let’s take a quick step back before we dive into it though.
What’s open source exactly?An open source project is a piece of source code that is publicly available for anyone under one of the many open source licenses out there. Someone, or a group of people from the same company or from disparate places, have “banded together” and created a piece of software that does something. They put the code of that software out in the open and slap a license on top of it. That ends up being an open source project.
Open source isn’t free. There’s a legal binding associated with using open source, but it isn’t what we’re interested in here. It is the fact that if you use open source, it doesn’t mean that you pay nothing to no one. It just means that you get *something* with no strings attached.
Why would anyone end up doing this for free? Well… that brings us to business models.
Open source business modelsThere are different types of open source licenses. Each with its own set of rules, and some more permissive than others, making them business-friendly. Sometimes the license type itself is used as a business model, simply by offering a dual license mode where a non-permissive open source license is available freely and a commercial one is available in parallel.
In other cases, the business model of the open source project revolves around offering support, maintenance and customization of that project. You get the code for free, but if you want help with it – you can pay!
Sometimes, the business model is around additional components (this is where you will see things like community edition and enterprise edition popping up as options in the project’s website). Things such as scripts for scaling the system, monitoring modules or other pieces of operational and functional components are protected as commercial products. The open source part brings companies to use it and raise popularity and awareness to the project, while the commercial one is the reason for doing it all. How the developers behind the project bring food to the table and become rich.
In recent years, you see business models revolving around managed services. The database is open source and free, but if you let us host it for you and pay for it, we’ll take care of all your maintenance and scaling headaches.
And some believe it is really and truly free. Troy Hunt wrote about it recently (it is a really good post – go read it):
“… there is a suggestion that those of us who create software and services must somehow be in it for the money”
To that I say – yes!
At the end of the day, delving into open source is all about the money.
Why?
- If you do this to create a popular project, then your aim is almost always to figure out how to monetize it. Directly (see above examples) or indirectly, by increasing your chances of getting hired for higher paying jobs or into more interesting projects
- Sometimes, you do this because you care deeply about a topic. But the end result is similar. You either have the time to deal with it because you make money elsewhere and this is a hobby – or because the company hiring you is HAPPY that you are doing it (which means you are doing it to some extent for the intrinsic value it gives you at that company)
- You might be doing it to hone your skills. But then again, the reason for all this is to become a better programmer and… get hired
The moment the open source project you are developing is meaningful to two more people, or even a single company, there are monetary benefits to be gleaned. We’d venture that if you aren’t making anything from these benefits (even minor ones), then the open source project has no real future. It gets to a point where it should either grow up or wither and die.
A few more words about open source projectsJust a few things before we start our journey to the WebRTC open source realm:
- Most open source projects are just an API abstracting out a certain activity or capability that you need for your own application development. In the case of WebRTC, we will be focusing on such abstractions that implement specific network entities – more on that later
- When using open source, you usually have a bit more control over your application. That’s because you can modify the source code of the open source components you use as opposed to asking from a vendor to do that when you use a precompiled library
- Many open source projects will have poor documentation. That will be doubly true when they are lacking a solid business model – hobbyists developers are more into writing code than they are explaining how to use that code
- Documentation is an important aspect for commercial use of open source projects. So are its ability to provide a clear API facade and code samples to make it easy to start using
A common mistake by “noobs” is that WebRTC is a solution that requires no coding. Since browsers already implement it, there’s nothing left to do. This can’t be farther away from the truth.
WebRTC as a protocol requires a set of moving parts, clients and servers; that together enable the rich set of communication solutions we’re seeing out there.
The diagram above, taken from the Advanced WebRTC Architecture course, shows the various components necessary in a typical WebRTC application:
- Clients, web-based or otherwise
- The web browser ones are the ones you get for “free” as part of the browser
- Anything else you need to figure on your own
- Application server, which we’re not going to touch in this article. The reason being that this is a generic component needed in any type of application and isn’t specific to WebRTC
- Signaling server, taking care of setting up and negotiating the WebRTC sessions themselves
- STUN/TURN server, which deals with NAT traversal. Needed in almost every deployment
- Media server, for media processing heavy lifting. Be it group calling, recording, video rendering, etc – a media server is more than likely to make that happen
For each and every component here, you can find one or more open source projects that you can use to implement it. Some are better than others. Many are long forgotten and decaying. A few are pure gold.
Lets dive into each of these components to see what’s available and at what state we find the open source community for them.
WebRTC open source client librariesFirst and foremost, we have the WebRTC open source client libraries. These are implementations of the WebRTC protocol from a user/device/client perspective. Consider these your low level API for WebRTC.
There used to be only a single one – libwebrtc – but with time, more were introduced and took their place in the ecosystem. Which is why we will start with libwebrtc:
libwebrtcTHE main open source project of WebRTC is libwebrtc.
Why?
- It is the first one to be introduced
- Chrome uses it for its WebRTC implementation
- The same goes for Safari, Edge and Firefox – each with a varying degree of integration and use
- Many of the native mobile apps use libwebrtc internally
Practically speaking – libwebrtc is everywhere WebRTC is.
Here are a few things you need to know about this library:
- libwebrtc is maintained and controlled solely by Google. Every change needs to be signed off by a Googler.
- It gets integrated into Chromium and Chrome, which means it reaches billions of devices
- That means that Google is quite protective about it. Getting a contribution into libwebrtc is no easy feat
- While there are others who contribute, external contributions to libwebrtc are rare and far between
- Remember also that the team at Google doing this isn’t philanthropic. It does that for Google’s own needs, which mostly means Google Meet these days. This means that use cases, scenarios, APIs and code flows that are used by Google Meet are likely to be more secure, stable and far more optimized than anything else in libwebrtc’s codebase
- Did we mention the whole build system of libwebrtc is geared towards compiling it into Chromium as opposed to other projects (like the one you’re building)? See Philipp’s Fosdem talk from 2021.
- Or that some of its interfaces (like device acquisition) are less tested simply because Chrome overrides them, so Google’s focus is on the Chrome interfaces and not the ones implemented in libwebrtc?
Looking at the contributions over time Google is doing more than 90% of the work:
The amount of changes has been decreasing year-over-year after peaking in early 2016. During the pandemic we even reached a low point with less than 200 commits per month on average. Even with these reduced numbers libwebrtc is the largest and most frequently updated project in the open source WebRTC ecosystem.
The number of external contributions is fairly low, below 10%. This doesn’t bode well for the future of libwebrtc as the industry’s standard library of WebRTC. It would be better if Google opened up a bit more for contributions that improve WebRTC or those that make it easier to use by others.
This leads us to the business model aspect of libwebrtc
Money time
What if one decides to use libwebrtc and integrate it directly in his own application?
- There’s no option for paid support
- No real alternative to pay for custom development
- Maintaining your own fork and keeping it in sync with the upstream one is a lot of effort
That said, for the most part, and in most situations, libwebrtc is the best alternative – that’s because it follows the exact implementations you will be bumping into in web browsers. It will always be the most up to date one available.
A side note – libwebrtc is implemented in C++. Why is this relevant? Pion
PionPion is a Go implementation of the WebRTC APIs. Sean DuBois is the heart and sole behind the Pion project and his enthusiasm about it is infectious.
Putting on Tsahi’s cynic hat, Pion’s success can be attributed a lot to it being written in Go. And that’s simply because many developers would rather use Go (modern, new, hip) and not touch C++.
Whatever the reason is, Pion has grown quite nicely since its inception and is now quite a popular WebRTC open source project. It is used in embedded devices, cloud based video rendering and recently even SFU and other media server implementations.
Money time
What if one decides to use Pion and integrate it directly in his own application?
- There’s no option for paid support
- No official alternative to pay for custom development
- There are a handful of contributors to Pion who are doing contracting work
There are other implementations of WebRTC in other languages.
The most notable ones:
- aiortc – a Python implementation of WebRTC
- WebRTC.rs – a Rust implementation of WebRTC, created as a rewrite of Pion
There are probably others, less known.
We won’t be doing any Money time section here. These projects are still too small. We haven’t seen too many services using them in production and at scale.
GStreamerGStreamer is an open source media framework that is older than WebRTC. It is used in many applications and services that use WebRTC, even without using its WebRTC capabilities (mainly since these were added later to GStreamer).
We see GStreamer used by vendors when they need to transform video content in real-time. Things like:
- Taking machine rendering (3D, screen casting or other) and passing them to a browser via WebRTC
- Mixing inputs combining them into a single recording or a single livestream
- Collecting media input on embedded platforms and preparing it for a WebRTC session
Since WebRTC was added as another output type in GStreamer, developers can use it directly as a broadcasting entity – one that doesn’t consume data but rather generates it.
GStreamer is a community effort and written in C. While it is used in many applications (commercial and otherwise), it lacks a robust commercial model. What does that mean?
Money time
What if one decides to use GStreamer and integrate it directly in his own application?
- There’s no official option for paid support
- No official alternative to pay for custom development
- The ecosystem is large enough to allow finding people with GStreamer knowledge
Next we have open source TURN servers. And here, life is “simple”. We’re mostly talking about coturn. There are a few other alternatives, but coturn is by far the most popular TURN server today (open source or otherwise).
In many ways, we don’t need more than that, because TURN is simple and a commodity when it comes to the code implementation itself (up to a point, as Cloudflare is or was trying to change that with their managed service).
But, and there’s always a but in these things, coturn needs to get updated and improved as well. Here’s a recent discussion posted as an issue on coturn’s github repo:
Is the project dead?
Read the whole thread there. It is interesting.
The maintainers of coturn are burned out, or just don’t have time for it (=they have a day job). For such a popular project, the end result was a volunteer or two from the industry picking up the torch and doing this in parallel to their own day job.
Which leads us to:
Money time
What if one decides to use coturn and integrate it directly in his own application?
- There’s no official option for paid support
- No official alternative to pay for custom development
- The ecosystem is large enough to allow finding people with coturn knowledge
Signaling servers are a different beast. WebRTC doesn’t define them exactly, but they are needed to pass the SDP messages and other signals between participants. There are several alternatives here when it comes to open source signaling solutions for WebRTC.
It should be noted that many of the signaling server alternatives in WebRTC offer purely peer communication capabilities, without the ability to interact with media servers. Some signaling servers will also process audio and video streams. How much they focus on the media side versus the signaling side will decide if we will be treating them here as signaling servers or media servers – it all boils down to their own focus and to the functions they end up offering.
Signaling requires two components – a signaling server and a client side library (usually lightweight, but not always).
We will start with the standardized ones – SIP & XMPP.
SIP and XMPPSIP and XMPP preceded WebRTC by a decade or so. They have their own ecosystem of open source projects, vendors and developers. They act as mature and scalable signaling servers, sometimes with extensions to support WebRTC-specific use-cases like creating authentication tokens for TURN servers.
We will not spend time explaining the alternatives here because of this.
Here, it is worthwhile mentioning MQTT as well. Facebook is known to be using it (at least in the past – not sure about today) in their Facebook Messenger for signaling
PeerJSPeerJS has been around for almost as long as WebRTC itself. For an extended period of that time, the codebase has not been maintained or updated to fit what browsers supported. Today, it seems to be kept.
The project seems to focus on a monolithic single server deployment, without any thought about horizontal scaling. For most, this should be enough.
Throughout the years, PeerJS has changed hands and maintainers, including earlier this year:
Without much ado, lets move to the beef of it:
Money time
What if one decides to use PeerJS and integrate it directly in his own application?
- There’s no official option for paid support
- No official alternative to pay for custom development
- The codebase is small, so if you know WebRTC, these challenges shouldn’t pose any real issue
Simple-Peer has been driven by Feross and his name in the early days. It is another one of those “pure WebRTC” libraries that focuses solely on peer-to-peer. If that fits your use-case, great, it is mature and “done”. Most of the time your use-case will evolve over time though.
It has received only a few maintenance commits in 2022 and not many more in 2021. The same considerations as for PeerJS apply for simple-peer. If you need to pick between the two… go for simple-peer, the code is a bit more idiomatic Javascript.
Money time
Just go read PeerJS – same rules apply here as well.
MatrixMatrix is “an open network for secure, decentralized communication”. There’s also an open standard to it as well as a commercial vendor behind it (Element).
Matrix is trying to fix SIP and XMPP by being newer and more modern. But the main benefit of Matrix is that it comes as client and server along with implementations that are close to what Slack does – network and UI included. It is also built with scale in mind, with a decentralized architecture and implementation.
Here we’re a bit unaligned… Tsahi thinks Matrix is a good alternative and choice while Philipp is… less thrilled. Their WebRTC story is a bit convoluted for some, meandering from full mesh to Jitsi to a “native SFU” only recently.
So… Matrix has a company behind it. But they have their own focus (messaging service competing with Slack with privacy in mind).
Money time
What if one decides to use Matrix and integrate it directly in his own application?
- There’s no official option for paid support
- No official alternative to pay for custom development
- That said, Matrix does have a jobs room on Matrix where you can search for paid help
At the time of writing, there are 26,121 repositories on github mentioning WebRTC. By the time you’ll be reading it, that number will grow some.
Not many are sticking out too much, and in that jumble, it is hard to figure out which projects are right for you. Especially if what you need needs to last. And doubly so if you’re looking for something that has decent enough support and a thriving community around it.
Open source SFUs and media servers in WebRTCAnother set of important open source WebRTC components are media servers and SFUs.
While signaling servers deal with peer communication of setting up the actual sessions, media servers are focused on the channels – the actual data that we want to be sending – audio and video streams, offering realtime video streaming and processing Whenever you’ll be needing group sessions, broadcasts or recordings (and you will, assuming you’d like video calls or video conferences incorporated in your application), you will end up with media servers.
Here’s where are are marketwise
Janus, Jitsi, mediasoup & PionI’ve written about these projects at length in my 2022 WebRTC trends article. Here’s a visual refresher of the relevant part of it:
Janus, Jitsi, mediasoup and Pion are all useful and popular in commercial solutions. Let’s try to analyze them with the same prism we did for the other WebRTC open source projects here.
Janus
- There’s official paid support available from meetecho
- You can pay meetecho for consulting and paid development. From experience, they are mostly busy which means they are picky with who they end up working with
- The Janus ecosystem is large enough and there are others who offer development services for it as well
Jitsi can be considered a platform of its own:
- At the heart of Jitsi is the Jitsi Videobridge, with additional components around it, composing together the Jitsi Meet video chat app
- There’s also a managed CPaaS service offering as part of it – 8×8 JaaS
Money time
- Jitsi was acquired a few years ago by 8×8. Which means that there’s no official option for paid support
- Similarly, custom development isn’t available
- The Jitsi ecosystem is large enough and there are others who offer development services for it as well
- Oh, and like Matrix (where Element offers paid hosting), 8×8 JaaS offers paid hosting for Jitsi (=CPaaS). There’s also Jitsi Meet which is essentially a free managed service built on top of Jitsi itself
- mediasoup is maintained by 2 developers who have a day job at Around. Which means that there’s no official option for paid support
- Similarly, custom development isn’t available
- The ecosystem around mediasoup means you can get developers for it as well
- We’ve already discussed Pion when we looked at WebRTC clients
- Assume the same is true for media servers
- Only you have the headache of choosing which media server written on top of Pion to use
–
To be clear – in all cases above, getting vendors to help you out who aren’t maintaining the specific media server codebase means results are going to be variable when it comes to the quality of the implementation. In other words, it is hard to figure out who to work with.
The demise of KurentoThe Kurento Media Server is dead. So much so that even the guys behind it went to build OpenVidu (below) and then made OpenVidu work on top of mediasoup.
Don’t touch it with a long stick.
It has been dead for years and from time to time people still try using it. Go figure.
Higher layers of abstractionA higher layer abstraction open source project strives to become a platform of sorts. Their main focus in the WebRTC ecosystem is to offer a layer of tooling on top of open source media servers. The two most notable ones are probably OpenVidu and LiveKit.video conferencing
OpenViduOpenVidu is a kind of an abstraction layer to implement a room service, UI included.
It originates from the team left behind from the Kurento acquisition. With time, they even adopted mediasoup as the media server they are using, putting Kurento aside for the most part.
Money time
Unlike many of the open source solutions we’ve seen so far, OpenVidu actually seem like they have a business model:
- There’s an official commercial support available
- There are hosted commercial plans available as well as consulting and development work
LiveKit offers an “open source WebRTC infrastructure” – the management layer above Pion SFU.
For the life of me though, I don’t understand what the business model is for LiveKit. They are a company – not just an open source project, and as such, they need to have revenue to survive.
Most probably they get some support and development money from enterprises adopting LiveKit, but that isn’t easily apparent from their website.
Other, less popular open source alternatives for WebRTCThere are other companies who offer commercial solutions that are proprietary in nature. Some do it as on premise alternatives, where they provide the software and the support, while you need to deploy and maintain.
These can either be suitable solutions or disasters waiting to happen. Especially when such a vendor decides to pivot or leave the market.
Tread carefully here.
Is it time for WebRTC open source to grow up?This has been a long overview, but I think we can all agree.
The current state of WebRTC open source is abysmal:
- We are more than 10 years in
- There are thriving open source projects for WebRTC out there
- These projects are used by many – hobbyists and professionals alike
- They are found inside commercial applications serving millions of users
- But they offer little in the way of support or paid help
- Somehow, the market hasn’t grown commercially
If it were up to us, and it isn’t, we’d like to see a more sophisticated market out there. One that gives more and better commercial solutions for enterprises and entrepreneurs alike.
The post The state of WebRTC open source projects appeared first on BlogGeek.me.
Be very clear to yourself why you manage your own TURN servers
Running your own TURN servers for your WebRTC application is not necessarily the best decision. Make sure you know why you’re doing it.
[In this list of short articles, I’ll be going over some WebRTC related quotes and try to explain them]
Are you running your own TURN server? Great!
Now, are you crystal clear and honest with yourself about why you’re doing that exactly?
WebRTC has lots of moving parts you need to take care of. Lots of WebRTC servers: The application. Signaling servers. Media servers. And yes – TURN servers.
I already covered a few aspects of TURN in this WebRTC quote – We TURNed to see a STUNning view of the ICE. It is now time to review the build vs buy decision around TURN.
You see, NAT traversal in WebRTC is done by using two different servers: STUN and TURN. STUN is practically free and it can also be wrapped right into the TURN server.
TURN servers are easy to interface with, but not as easy to install, configure and maintain properly. Which is why my suggestion more often than not is to use a third party managed TURN service instead of putting up your own. Economies of scale along with focus and core competencies come to mind here with this decision.
Why buy your WebRTC TURN servers?Buying a TURN server should be your default decision. It is simple. It isn’t too expensive (for the most part) and it will reduce a lot of your headaches.
Most of the companies that approach me with connectivity issues of their WebRTC application end up in that state simply because they decided to figure out NAT traversal in WebRTC on their own.
Here are a few really good reasons why you should buy your TURN service:
- The best practices of TURN (and STUN) configuration aren’t the defaults of open source TURN servers or of the standard specification itself. So if you don’t have someone inhouse who has done it at scale in the past already, then don’t start now
- Using a third party managed TURN server is simple. Onboarding and integration should be a breeze (a few hours at most)
- There’s no real vendor lock-in. Switching to your own TURN servers will cost you the same as it would to start with your own TURN servers, so you can delay that decision for later. And switching to another managed TURN server is just as simple as it is to start using one for the first time
- Testing for edge cases and figuring out issues with WebRTC connectivity is hard. It takes a lot of time, requires patience, understanding and visibility when issues fail. None of this is something you’ll have in the first months of running your own service
- It is cheap. Twilio has it at $0.4/gigabyte of data. And not all of your traffic will go through TURN anyways. When you’ll start paying too much to your taste, you will be able to put up your own infrastructure. But why invest in that effort before it is time to do so?
- Someone else will take care of scaling. TURN needs to be as close as possible to the end users. Installing a single server won’t be enough. Installing a single region won’t be enough. Why deal with that headache?
- Firewall friendliness. Using your own servers means opening them up in firewall configurations of your customers. There’s a small likelihood that these firewalls are already configured to support the managed TURN service you are using for other tools
We are all builders. And we love building. So adding TURN into our belt of things we built makes sense. It also plays well into the vertical integration we now appreciate with how successful Apple has been with it with its services.
But frankly, it is mostly about control. The ability to control your own destiny without relying on others.
I still think you should buy your TURN servers from a reputable managed service provider. That said, here are some good reasons why to build and deploy your own:
- Data sovereignty and other regulatory reasons. In some industries, for some customers, the fact that you host and run your own servers is critical. In such a case, using a managed third party TURN service is simply impossible. In the same domain, privacy and data processing requirements may make using a third party harder than setting up your own
- You already have a large traffic and footprint. With economies of scale this starts becoming interesting and important. If you have the sheer size that makes it worthwhile running your own then do it. I wouldn’t start below $10,000 or even $50,000 in monthly expenses for your managed TURN service, which is a lot of traffic. Why? Because you’ll need a full time ops person on the job for at least half a year if not longer. And you’ll need to deploy servers in many regions from the get go, so better start when you’re big enough
- Firewall configurations can be a mess. Sometimes, your customers may want to validate the IP addresses they configure are yours, or want to limit the IP address ranges they configure, or limit the services they expose themselves to. In such cases, they might not look at it nicely when you use a third party
- Existing customer installations might already be configured to your IP address ranges, and just placing your TURN servers within those ranges will be easier than asking them to change firewall configurations to incorporate a third party vendor
- Traffic control is another reason. Using your own SDN network configuration or packet acceleration may benefit from having your own TURN servers in-house, alongside the rest of your infrastructure as opposed to be hosted elsewhere where connectivity to your backend servers might be questionable
–
Build? Buy? Which one is the path you’ll be taking?
Trying to get more of your calls connected in WebRTC? Check out this free video mini course on effectively connecting WebRTC sessions
The post Be very clear to yourself why you manage your own TURN servers appeared first on BlogGeek.me.
We TURNed to see a STUNning view of the ICE
Every time you look at NAT Traversal in WebRTC, you end up learning something new about STUN, TURN and/or ICE.
[In this list of short articles, I’ll be going over some WebRTC related quotes and try to explain them]
STUN, TURN and ICE. The most misunderstood aspects of WebRTC, and the most important ones to get more calls connected. It is no wonder that the most viewed and starred lesson in my WebRTC training courses is the one about NAT traversal.
Let’s take this opportunity to go over a few aspects of NAT traversal in WebRTC:
- STUN is great (and mostly free). It doesn’t route media, it just punches holes in firewalls and NATs
- TURN means relaying your media. It isn’t used for all sessions, but when it is used, it is a life saver for that session. You can keep the TURN servers on all connections, since it will be used only when needed
- While STUN and TURN are servers, ICE isn’t. ICE is a protocol. It is how WebRTC decides if it is going to use TURN or not in a session
- No matter how you connect your session, it may happen on either UDP or TCP. UDP will be a better alternative (and WebRTC will prioritize it and try to connect it “first”)
- TURN servers are expensive. Don’t use free TURN servers – they aren’t worth the money you aren’t paying for it. Use your own or go for a paid, managed TURN service
- Put TURN servers as close as possible to your users. They’ll thank you for that
- In the peer connection’s iceServers configuration – don’t put more than 3-4 servers (that means 1 STUN, 1 TURN/UDP, 1 TURN/TCP, 1 TURN/TLS). More servers means more connectivity checks and more time until you get things connected – it doesn’t mean better connectivity
- Geolocation with TURN should be done either before you place your TURN servers in the configuration or via the DNS requests for the TURN servers themselves
- You don’t always need TURN servers. Read more about when you need and don’t need TURN
This covers the basics. There’s a ton more to learn and understand about NAT traversal in WebRTC. I’d also suggest not installing and deploying your own TURN servers but rather use a third party paid managed service. The worst that can happen is that you’ll install and run your own later on – there’s almost no vendor lock-in for such a service anyway.
Trying to get more of your calls connected in WebRTC? Check out this free video mini course on effectively connecting WebRTC sessions
The post We TURNed to see a STUNning view of the ICE appeared first on BlogGeek.me.
With media delivery, you can optimize for quality or latency. Not both
You will need to decide what is more important for you – quality or latency. Trying to optimize for both is bound to fail miserably.
[In this list of short articles, I’ll be going over some WebRTC related quotes and try to explain them]
First thing I ask people who want to use WebRTC for a live streaming service is:
What do you mean by live?
This is a fundamental question and a critical one.
If you search Google, you will see vendors stating that good latency for live streaming is below 15 seconds. It might be good, but it is quite crappy if you are watching a live soccer game and your neighbors who saw the goal taking place 15 seconds before you did are shouting.
I like using the diagram above to show the differences in latencies by different protocols.
WebRTC leaves all other standards based protocols in the dust. It is the only true sub-second latency streaming protocol. It doesn’t mean that it is superior – just that it has been optimized for latency. And in order to do that, it sacrifices quality.
How?
But not retransmitting or buffering.
With all other protocols, you are mostly going to run over HTTPS or TCP. And all other protocols heavily rely on retransmissions in order to get the complete media stream. Here’s why:
- Networks are finicky, and in most cases, that means you will be dealing with packet losses
- You stream a media file over the internet, and on the receiving end, parts of that file will be missing – lost in transmission
- So you manage it by retransmission mechanisms. Easiest way to do that is by relying on HTTPS – the main transport protocol used by browsers anyways
- And HTTPS leans on TCP to offer reliability of data transmission, which in turn is done by retransmitting lost packets
- Retransmissions require time, which means adding a buffering mechanism to make room for it to work and provide a smooth viewing experience. That time is the latency we see ranging from 2 seconds up to 30 seconds or more
WebRTC comes from the real time, interactive, conversational domain. There, even a second of delay is too long to wait – it breaks the experience of a conversation. So in WebRTC, the leading approach to dealing with packet losses isn’t retransmission, but rather concealment. What WebRTC does is it tries to conceal packet losses and also make sure there are as little of it as possible by providing a finely tuned bandwidth estimation mechanism.
Looking at WebRTC itself, it includes a jitter buffer implementation. The jitter buffer is in charge of delaying playout of incoming media. This is done to assist with network jitter, offering smoother playback. And it is also used to implement lip synchronization between incoming audio and video streams. You can to some extent control it by instructing it not to delay playout. This will again hurt the quality and improve latency.
You see, the lower the latency you want, the bigger the technical headaches you will need to deal with in order to maintain high quality. Which in turn means that whenever you want to reduce latency, you are going to pay in complexity and also in the quality you will be delivering. One way or another, there’s a choice being made here.
Looking to learn more on how to use WebRTC technology to build your solution? We’ve got WebRTC training courses just for that!
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Nocode/Lowcode in CPaaS
Lowcode and nocode or old/new concepts that are now finding their way to Communication APIs. Here’s the latest developments.
Lowcode and nocode has fascinated me. Around 15 years ago (or more), I was tasked with bringing the video calling software SDKs we’ve developed at RADVISION to the cloud.
At the time, the solutions we had were geared towards developers and were essentially SDKs that were used as the video communication engines of applications our customers developed. Migrating to the cloud when all you are doing is the SDKs is a challenge. How do you offer your developer customers with the means to control the edge devices via the cloud, and doing so while allowing the application to control the look and feel, embedding the solution wherever they want.
The cloud we’ve developed used Python (Node.js wasn’t popular yet), and we dabbled and experimented with Awesomium – a web browser framework for applications – the predecessor of today’s more popular Electron. We built REST APIs to control the calling logic and handle the client apps remotely via the cloud.
I spent much of my time trying to come to grips with how exactly you would fit remote controlling an app to the fact that you don’t really own or… control. A conundrum.
Fast forward to today, where cloud and WebRTC are everywhere, and you ask yourself – how do you remote control communications – and how do you build such interactions with ease.
The answer to that is usually by way of nocode and lowcode. Mechanisms that reduce the amount of code developers need to write to use certain technologies – in our case Communication APIs (CPaaS).
I had a bit of spare time recently, so I decided to spend it on capturing today’s nocode & lowcode status and progress within the CPaaS domain.
This has been especially important if you consider the recent announcements in the market – including the one coming from Zoom about their Jumpstart program:
“With Jumpstart, you can quickly create easy-to-integrate and easy-to-customize Zoom video solutions into your apps at lower costs.”
So without much ado, if this space interest you, you should check out my new free eBook: Lowcode & Nocode in Communication APIs
This eBook details and explains the various approaches in which lowcode and nocode manifest themselves in the Communication APIs domain. It looks into the advantages and challenges of developers who adopt such techniques within their applications.
I’d like to thank Daily for sponsoring this ebook and helping me make it happen. If you don’t know them by now then you should. Daily offers WebRTC video and audio for every developer – they are a CPaaS vendor with a great lowcode/nocode solution called Daily Prebuilt
If you are in the process of developing applications that use 3rd party Communication APIs, you will find the insights in this eBook important to follow.
GET MY FREE LOWCODE/NOCODE CPAAS EBOOKThe post Nocode/Lowcode in CPaaS appeared first on BlogGeek.me.
In group video calls, effectively managing bandwidth is 90% of the battle
The biggest challenge you will have when implementing WebRTC group calling is estimating optimizing bandwidth use.
[In this list of short articles, I’ll be going over some WebRTC related quotes and try to explain them]
Video is a resource hog. Some say that WebRTC is a great solution for 1:1 calls, but is lacking when it comes to group calling. To them I’d say that WebRTC is a technology and not a solution. In this case, it simply means that you need to invest some effort in getting group video calling to work well.
What does that mean exactly? That you need to think about bandwidth management first and foremost.
Why?
Let’s assume a 25 participants video call. And we’re modest – we just want each to encode his video at 500kbps – reasonable if we plan on having everyone at a mere VGA resolution (640×480 pixels).
Want to do the math together?
We end up with 12.5Mbps. That’s only for the video, without the overhead of headers or audio. Since we only need to receive media from 24 participants, we can “round” this down to 12Mbps.
I am sure you have a downlink higher than 12Mbps, but let me tell you a few things you might not be aware of:
- A downlink of 100Mbps doesn’t mean you can really get sustainable 12Mbps for a long period of time
- It also doesn’t mean you can get 12Mbps of incoming UDP traffic (and you prefer UDP since it is better for sending real-time media)
- Most likely, your device won’t be able to decode 12Mbps of video content at reasonable CPU use
- And if you have hardware acceleration for video decoding, it usually is limited to 3 or 4 media streams, so handling 24 such streams means software decoding – again running against the CPU processing limit
- The larger the group the more diverse the devices and network connections. So you’ll be having people joining on old devices and smartphones, or with poor network connections. For them, 12Mbps will be science fiction at best
- As a rule of thumb, I’d look at any service that uses over 3-4Mbps of downlink video traffic for video group calls as something that wasn’t properly optimized
You can get better at it, trying to figure out lower bitrates, limit how much you send and receive and do so individually per participant in the video group meeting. You can take into consideration the display layout, the dominant speaker and contributing participants, etc.
That’s exactly what 90% of your battle here is going to be – effectively managing bandwidth.
Going for a group video calling route? Be sure to save considerable time and resources for optimization work on bandwidth estimation and management. Oh – and you are going to need to do that continuously. Because WebRTC is a marathon not a sprint
Scaling WebRTC is no simple task. There are a lot of best practices, tips and tricks that you should be aware of. My WebRTC Scaling eBooks Bundle can assist you in figuring out what more you can do to improve the quality and stability of your group video calling service.
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