Spearline acquired testRTC and now supports WebRTC testing and monitoring. This will change what I do, but in good ways.
For starters – I am still going to do what I did so far – have fun and help companies with their WebRTC and CPaaS challenges.
I tried to keep testRTC at an arm’s length from BlogGeek.me and what I do here just because… well… not sure why. Probably to stay as impartial as I can with the things that I do. That said, it is probably a good time to explain where we are with testRTC and our support for WebRTC applications.Where are we with testRTC?
We’ve started testRTC with the intent of providing a self service, cloud hosted testing solution for those developing with WebRTC. Along the way, we’ve expanded our product lines to include 3 separate domains with 5 different products:
Simply put, we are the only vendor today offering support for the full lifecycle of your WebRTC application – from development to deployment and long term maintenance of the service. We do that at scale, in the cloud, with a big smile
And then we met Spearline, and found a common ground.Who and what is Spearline?
Spearline offers testing and monitoring for your telephony services.
They have a large global deployment with real phone numbers across 70+ countries and carriers worldwide – landline and mobile. If you need your phone numbers tested and validated for their quality and performance (and you do), then you go to Spearline. Why? Because without actually testing a number, your only insight that a number isn’t working (say your sales line) is to get a customer to complain about it – which is way too late.
This all made perfect sense for us at testRTC. When we were approached, it was easy to figure out that this falls into this category:
We’re completing Spearline in a few ways (WebRTC being an important part of it), and Spearline completing testRTC in other ways (telephony, scale and enterprise sales to give a few of the things we were after).
Which leads me to rocket surgery.Rocket surgery
I had a technical call the other day. Related to BlogGeek.me. Someone at the call said “rocket surgery” at some point. It took me a few seconds to deconstruct that and understand it – he probably meant to say rocket science or brain surgery – just to indicate that they’re doing things that are hard, but not that hard (he said “this isn’t rocket surgery”).
Then it dawned on me. Rocket surgery is the best term I have for what we’re currently doing.
We’re marrying the best of both worlds here at testRTC & Spearline, so we can now offer our customers rocket surgery solutions. Things that no other vendor out there can do for you.
And that excites me – the things we can achieve and the plans we’re making for the future as part of this acquisition.What changes for BlogGeek.me?
Nothing and everything.
(can you spot the 10 differences between the images above?)
I am continuing my work at testRTC as before. Not as CEO (never liked that role), but as head of products for testRTC (which is kinda like a small CEO). testRTC is my baby. I want to see it grow and flourish.
But then again, I like the diversity and the thrill and fun of doing everything. And Spearline were kind enough to allow me to continue with my extra curricular activities. These include the courses, the weekly, insights, consulting and Kranky Geek.
I’ve been thinking a lot lately about my future. And what else I want to do. I don’t have the answers to it yet. For the foreseeable future though, this is going to be helping you with your WebRTC and CPaaS needs.Onward and upward
2021 has been a rollercoaster. I enjoyed the ride.
Here’s for a 2022 that is thrilling, exhilarating and fun.
The post Spearline acquiring testRTC – this is rocket surgery appeared first on BlogGeek.me.
There is a cool new feature everyone has been trying to implement – background transparency. Virtual backgrounds have been around for a while. Rather than inserting a new background behind user(s), transparency removes the background altogether, allowing the app to place users over a screen share or together in a shared environment. There doesn’t seem […]
The post How to add virtual background transparency in WebRTC appeared first on webrtcHacks.
Twilio Signal 2021 defines Twilio as “API”, “programmable”, “platform” and “customer engagement”. Here’s how it intends to compete in its many markets.
Twilio Signal 2021 is when Twilio officially pivoted from CPaaS to a Customer Engagement Platform. This is the reason Twilio acquired Segment last year, and the explanation of how it intends to leverage that acquisition.
Every year, I put time aside for Twilio Signal. Either in person or remote, going through the sessions and paying extra attention during the keynote. This has developed into a comprehensive view and research resources about Twilio that I’ve put up. It is time now to review what we had at Twilio Signal 2021.Table of contents
Twilio didn’t put the keynote for Signal 2021 on YouTube (yet), but they did have it as part of their all-day Signal TV session. The video below will get you the keynote, which was around 90 minutes long:
As events go, Twilio Signal 2021 was quite a good experience for a virtual event. It was a bit hybrid, but most of the focus and action took place on the virtual side of it (or at least felt that way for me as a virtual audience).Defining Twilio in 2021
Twilio never liked or used the term CPaaS. I am not really sure why.The Twilio pivot
There were 4 words that came time and time again during the keynote, and I think they are the center of what Twilio gravitates around today: “programmable”, “platform” and “customer engagement”.
Everything Twilio does can be found around these words, and I believe also every type of adjacent business they will try to go after will have two or more of these words in them in one way or another.
Twilio tried to show this shift and to move away a bit from APIs. It will take more than a single Signal event to do that.
Jeff Lawson, Co-founder and CEO of Twilio, started by presenting the idea of Customer Engagement and ended the keynote with the Customer Engagement Platform taking us in a complete circle around it.Why did Twilio pivot now?
Twilio is the leader in CPaaS. It has been so for many years now, defining and redefining what CPaaS is. Twilio is also ahead of all of its competitors. Way ahead. It acts as a best of suite provider, which covers most if not all of what CPaaS is, with depth of functionality in many of its offerings.
As such, it sees and knows the market. It also knows the market’s limits. Which means it understands its estimated growth. It had to pivot and start eating up more adjacencies to continue growing at an accelerated rate. But there probably aren’t enough adjacencies it can go after that can be defined as CPaaS or as communication APIs. So they went up the food chain, marketing customer engagement as their target.How Twilio’s breakout acquisitions into email and customer data enabled the pivot to Customer Engagement
Twilio’s reasoning for doing it now?
To be frank, the architectura shift as well as the move from reactive to proactive have been industry themes for over 10 years. The pandemic simply accelerated these changes, and probably accelerated Twilio’s own pivot. It is also a new language that Twilio is now speaking, so we hear it from them as well.Twilio by the numbers
Each time, Jeff starts his keynote with numbers, showing off Twilio’s size. It is interesting each time to see which numbers he shares and highlights at the beginning of the keynote. This year?Twilio Signal 2021 numbers versus 2019 & 2020
What numbers did Twilio share in the beginning of its keynote this year versus previous years?201920202021Customers160,000200,000+240,000+ in 180+ countriesText messages––128B (100% growth)Emails––1T (5.8B single day peak)Calls––25BFlex interactions––0.5BSegment data events––10TInteractions750B1T–Unique phone numbers2.8B3B–Calls/minute32,500––Peak SMS/second13,000––Email addresses3B/quarter50%–Video minutes–3B–Developers6M––
This is in-line with its pivot, as many of the original numbers aren’t even mentioned.
So… Twilio is now even bigger, and it is pivoting.
I haven’t added the social good related numbers that Twilio shared not because they aren’t important, but because they require a separate mention.
Twilio made the decision years ago to be a company that does good in the world. It also decided to put its money where its mouth is, through its twilio.org operation and its shift to become a diversified company.
Time is spent each year at Signal during the keynotes as well as in specific sessions for social good, and this year was no different.Twilio and partnerships
Jeff mentioned the strategic partners of Twilio at the beginning as well. These are getting more important to Twilio as it grows and shifts towards customer engagement.Twilio dogfooding
Twilio is dogfooding its own products. For Twilio Signal 2020 and 2021 it has been hard at work building its own hybrid events platform. Still at its early stages but quite commendable.
Each year, additional pieces of the Twilio building blocks are being used to create these events. It will be interesting to see if in 2022 they will continue with this trend or go to a live-only event. Another question is if and when will they productize this as a programmable events platform.The Pivot: Twilio Customer Engagement Platform
After the numbers it was time for the pivot. This is where Twilio moved away a bit from its roots into communications towards custom engagement. And the way this is explained by the fact that Twilio now isn’t only about communications but about all experiences with customers. Customers “drove” Twilio there, which led to the creation of Twilio’s Customer Engagement Platform.Setting the stage
Two things here:
If you look at the communications market diagram above which I like using, then Twilio encompasses two of the three domains. The difference now is that it is vying towards the CRM part with its new story of a customer engagement platform.
The pillars of Twilio’s Customer Engagement Platform?
From here on, the keynote was focused on showcasing everything revolving around customer experience with trust, scale, reliability and compliance as the main themes.FUDing the enterprise
To hammer the message through, Twilio decided to harness the “digital giants”. In its mind, these are Amazon, Google, Netflix and Facebook. An odd choice, as Apple and Microsoft would be “gianter” than Netflix…
The reason behind this, is that these companies make the best use of customer data to improve its engagement with its customers, providing a singular, cohesive view of them.
Logic states that these digital giants have grown with the pandemic because they understand their customers better, and other vendors need to follow suit or be gobbled up by these digital giants.
Now that we want to be like them, we need to have the technology to do that. Amazon didn’t buy its CRM from anyone, it built it. It fed it with the data needed. And so do you dear vendor – you can’t rely on an existing CRM – you will need to build it. And just accidentally, Twilio Flex is what you need to build it (wink wink ).
Oh, but it isn’t Twilio Flex. It is actually Twilio Flex + Segment + machine learning.
To hammer that in, Jeff made sure you know that you don’t want the digital giants as your partners when it comes to your customers: Amazon taking a cut of each purchase,the Apple tax, Facebook and Google auctioning user attention via ads. You dear vendor, need and want to own your customer relationship – directly:
Now that we’re all warmed up, it was time to share and explain what Twilio Customer Engagement Platform really is.The Twilio Customer Engagement Platform Twilio’s new Marketecture: Twilio Customer Engagement Platform
Twilio’s new Marketecture: Twilio Customer Engagement Platform
Jeff went through the platform’s components, which sits well with its current set of product offerings and acquisitions.1. Channels
Channels are the basic Twilio building blocks. That’s roughly the CPaaS part of Twilio:
The purpose is to be where the customer is.
Messaging and Voice is what Twilio is focused on. Ads were not mentioned anywhere else. Email is the SendGrid acquisition. And Video… well… that’s almost the only place it appeared during the keynote (more on video later).2. Engagement Apps
These are the higher level programmable applications that Twilio is offering:
This is why Twilio acquired Segment a year ago, and this is where it is taking Segment next.
The reason behind acquiring Segment was to pivot towards customer engagement and provide a larger offering to larger enterprises.
As Jeff said it, this is about engaging customers in real time at scale – that’s the focus of Segment.
From here, the keynote went to specific product announcements.Twilio Signal 2021 keynote announcements
During the keynote, several official announcements were made. There were others that didn’t make it into the keynote itself, which goes to show where the main focus is.
Here are the things announced in the keynote:
Jeff introduced this first and explained that this was their biggest architectural change.
Twilio switched from a single US based data center to enabling running the Twilio stack from multiple regions. A customer can potentially choose where he wants to connect to Twilio and where he wants his data to reside.
The main difference is lower latency on API calls if sent to the same region, but mainly the ability to choose where to run and store the data.
The actual deployment of this is going to happen in stages with a growing number of locations as well as products enabled. This will start with two new regions – Australia and Ireland, to cover Europe and Asia by year end for Twilio Voice; while Twilio Segment can store data in Europe.
The main reason for this is the growing need to support regional data storage to meet regulation in different countries and the need to entice larger enterprises to use Twilio.
This was announced before the explanation of the Customer Engagement Platform, but I decided to place it here, as part of the announcements of the keynote.Twilio MessagingX
The first announcement after introducing Twilio Customer Engagement Platform was Twilio MessagingX – the Channels layer in the new marketecture. This is also where the heart of the Twilio CPaaS solution lies.
It started nice. Soumya Srinagesh, Twilio’s VP Messaging Exchange, shared her big number:
Somehow, it differed from Jeff’s by 28B. I am sure there’s a good explanation, though either way, 100B is a large enough number.SMS centered, but evolving
For Twilio, messages are still SMS. It wasn’t said out loud, but it was hinted strongly enough throughout the session based on the announcement and in the analysts briefing for Twilio MessagingX:
During the analyst briefings of Twilio Signal 2021 the above slide was shared. I like it because it says a lot about how Twilio sees things in the messaging space. I also like it because of the way things are arranged.
Here are my immediate insights from it:
So what exactly is Twilio MessagingX?
It looks at messaging not from the API building block level, but rather from 3 different perspectives, each with its own set of focus and investments: Trust, Quality and Choice.
To be clear, all CPaaS vendors strive to do that. Twilio is one of the few that are big enough with economies of scale to really deliver it, and do so with programmability in mind in all of the possible layers.Trust
To handle trust, mainly deliverability and compliance, Twilio announced TrustHub.
TrustHub is all about compliant phone numbers (did we say SMS?)
It isn’t as if other CPaaS vendors don’t offer compliant phone numbers. TrustHub does that by enabling access to it via APIs as well, making it… programmable? More flexible?
The intent at the end of the day here is to have messages pass unfiltered and not get them to be blocked by carriers. Especially now, when our phone’s spam folders for SMS and voice are full of such numbers and messages.
This initiative is starting with the US market and will expand elsewhere.Quality
This is about deliverability by selecting which carriers to use to route messages, and figuring out bad connections. Twilio does that proactively (other CPaaS vendors do or say they do as well).
Not much else was said about it during the keynote, but this is where many of its acquisitions and investments in communication providers such as Syniverse earlier this year come to play.
This is a topic for a separate future analysis though.Choice
Choice is omni-channel. The ability to send messages to users on the channels they prefer.
There were two announcements around choice that were made:1. Google Business Messages
Twilio already had SMS, Facebook Messenger and Whatsapp. Now they added support for Google Business Messages – the ability of customers to start a conversation with a business directly from a Google search result or a map listing.
Interestingly, Twilio still has no Apple Business Chat support. Probably because Apple doesn’t want to deal with generic CPaaS vendors just yet.2. Content API
To manage and handle the fact that each messaging channel has slightly different rules you need to deal with, the new Twilio Content API is there to allow writing a message once and delivering it on whatever channel, with Twilio taking the headache of matching the message you want to send to how each channel likes that message.
As messages become more complex, requiring the user to take actions for example, such an API becomes a nice add-on.
For the most part, it feels like a utility that reduces a lot of the headache of a developer.Twilio Voice and IVR Now
This was the first time voice was discussed. It was preceded by this nice number:
We had 25B calls, now with 36B voice minutes. If both relate to voice, then that’s 1:26 minutes per call on average. Transactional is the main focus of Twilio.
Not much more has been said or announced about Twilio Voice directly. The only thing was IVR Now, with about a minute spent on explaining it:
IVR Now seems to be a program that is designed to assist enterprises to migrate their VoiceXML from on premise IVRs to Twilio’s IVR. If I had to guess, this is about offering professional services either by Twilio directly or via partners.
The reason for sharing this during the keynote was to get enterprises listening in to talk to Twilio about it – there still isn’t anything on Twilio’s website about this program…
Other than that, it felt out of touch with the rest of the keynote.Twilio Intelligence
Al Cook, VP & GM, Artificial Intelligence was the one introducing Twilio Intelligence. Al was the one leading and announcing Twilio Flex a few years ago, and this in a way is an extension of it.
The premise of Twilio Intelligence is the need to get from voice to data to meaning.
Twilio Autopilot was released to beta in 2018 and GA’d during Twilio Signal 2019. Interestingly, this is a platform and not a product (which means it probably is still Twilio Autopilot).
What is included?
A view of the language operators of Twilio Intelligence as implemented as part of Twilio Flex
Here’s what it means that Twilio Intelligence is a platform:
The demo was quite interesting, so I decided to share the direct pointer to it in the keynote here, as that’s easier than explaining it:
What I think:
It is hard work, and it will be interesting to see if Twilio nailed it this time around and what the next iteration of this will look like.
Where and when?
Now in limited private beta. A broader private beta in early 2022.
English only for now. Voice based for now.Twilio Flex
Twilio Flex launched 3 years ago. At the time, it was questioned if this would be successful or not. To some extent, it still is. The interesting thing is that the same was said about Amazon Connect, which took about 3 years to mature enough to show its size in the market.
Sateja Parulekar, Head of Contact Center Solutions at Twilio made it a point to explain that:
There were new announcements around Flex, mainly Flex ONE and Flextensions.Flex ONE
Flex ONE is about adding new channels to the Flex contact center with a single API. That includes today voice, messaging (including Whatsapp), chat and email.
The end result is one page holding all conversations across all channels with the customer.Flextensions
Flextensions are pre-build extensions to Twilio Flex. To me it sounded much like Zoom Apps or application directories of other enterprise tools.
This is geared on top of the partnerships that Twilio has been working hard on and explained in last year’s Signal 2020 when they discussed the Twilio Flex ecosystem. It is the right move for the Flex platform.
From a product perspective, the future of Flex lies in its integration with Segment. This is where Twilio Intelligence is most focused on, as we’ve seen in its introduction and demo.Segment
Peter Reinhardt, GM of Twilio Segment came to explain two things:
Segment is about collecting customer data from multiple sources and making it available as the single source of truth to wherever the business needs that data – all in real time.
Businesses store data about customers in many different places. With the migration towards cloud and SaaS, the number of these places is growing fast. I know… my own small business to run this website and my courses have their own share of SaaS vendors that I am using, all cobbled up with half-made integration and knit together with this masking tape called Zapier. It works. For my single person small business. Somewhat (I have tons of things I’d love to have better integrated, but don’t have the time or inclination to do – not enough ROI in it).
For real businesses, not like mine, the problem is a lot bigger and a lot more important to solve. Especially if… you want to be like the digital giants Jeff talked about at the beginning of the keynote and Peter made sure you remembered.
But back to the why:
And we’ve already seen glimpses of it with Twilio Intelligence earlier on.
I think Segment was the most interesting acquisition of Twilio so far. It isn’t only closing a gap on something they don’t have or need. It isn’t even going after a close adjacency. It is about being able to double down on customer engagement… and building a platform for it.
Which is exactly where Jeff started and where the keynote ends.Twilio Engage
Twilio Engage was the last announcement. This is the new engagement app that Twilio decided to launch. Flex is for support, Frontline is for sales and Engage is for marketers. This is the marketing cloud offering of Twilio, built on top of Segment.
It is available in pilot now and as GA in Q1 next year.
Not much else was explained or shared about this and the demo was mostly a concept of what can be done with it. Next year’s Signal event will probably show the flashy UI Peter said was less important than the dataAnnouncements that didn’t make it into the keynote
Video. IoT. Frontline. Sendgrid.
Probably a few others that I missed.
I’d like to discuss 2 of these announcements here in brief.Twilio Video Insights
Video isn’t (and was never) top of mind for Twilio. They have it supported, but somehow it feels like a second class citizen most of the time: Twilio WebRTC Go was announced in Signal 2020 to give a semblance of progress with video. It is a free peer-to-peer video service from Twilio that is limited in scale. It got some increased capacity this year especially for Signal 2021. Nothing to write home about (I already discussed these free WebRTC video APIs at length recently.
What was announced was Twilio Video Insights and Twilio Video, both very different from each other.
Twilio Video Insights collects WebRTC and other statistics off of your calls done over Twilio Programmable Video, to create a dashboard view of media quality.
This is similar to what we do at testRTC with our watchRTC product.
A demo was shown in one of the sessions of Twilio Signal.
For me this validates our own watchRTC product, as Twilio saw the need to offer that out of the bex as part of its service. That said, if you need something like this (for Twilio, another CPaaS vendor or your own infrastructure), then come check for yourself which tool is most suitable for your needs.Twilio Live
Twilio Live was announced a bit prior to Signal 2021. Probably in order to give center stage to Twilio Customer Engagement Platform where Live (or video for that matter) play a marginal role if any.
Here’s what I learned about Twilio Live during Signal 2021:
It is an interesting route that Twilio took for its broadcasting service. I am not sure how well it can compete with other CPaaS vendors who are clocking 100s of users or more per single WebRTC session. And it is hard to see this as an alternative for those using CDN streaming services already.
What will be interesting to see is how vendors accept this product and its position in the market – will this be good enough or even perfect for certain customers that can’t find the right solution for their broadcasting needs elsewhere.What Twilio isn’t
After writing down this longform article and analysis of Twilio Signal 2021, I think the most important part is what wasn’t said. And that’s what Twilio isn’t.
I long suggested and thought that CPaaS, CCaaS and UCaaS are going to merge as the lines between them are blurring. Vendors in each of these segments are vying towards the others through new product announcements and acquisitions.
Twilio went after CCaaS with Flex. It only made sense it would move into UCaaS at some point, being a comfortable adjacency in communications.
But it didn’t.
It went after customer engagement. Acquired Segment and doubled down in this route – making a splashing announcement of it at this Signal event and keynote.
Twilio is all about businesses communicating with customers.
Twilio is a lot less about people collaborating with each other in a business. Why? Because that’s where the focus of UCaaS is, and a lot of that focus relies on a slightly different set of requirements and roadmap.
This is also why video is getting less attention by Twilio for example.What’s next for Twilio?
I don’t really know.
This can be seen as a pivot, but also as the next step in Twilio’s evolution.
Twilio is surprising with the way it handles itself in the market, at least for me.
If I had to bet, I’d say that the next 2-3 years are going to be more of the same. Twilio will work on its current set of engagement applications, pouring data from the Segment CDP into it, and fitting its solutions for sales, support and marketing. Obviously, developers are still an important part of all of this.
I wouldn’t expect Twilio to go into additional adjacencies in the API domain or to go after unified communication related use cases either. At least not now. They have their hands full going up market and out of their comfort zone of pure communications.
The post Twilio Signal 2021: A Pivot from CPaaS to Customer Engagement Platform appeared first on BlogGeek.me.
WebRTC insights is turning out to be fun to create and super useful to our clients, looking to navigate the world of WebRTC.
Philipp Hancke and myself started this new thing called WebRTC Insights a year ago. We work well together, so we simply searched what we can do other than the WebRTC codelab, which was and still is a fun project.
WebRTC Insights is meant to help vendors sift through the technical (and non-technical) information that is out there and ever changing around WebRTC. Anything from bugs found, important changes in the WebRTC implementation to security issues raised and many other topics.
The idea? If you are a developer who uses WebRTC on a daily basis and relies on it, we can reduce the time you spend on finding what can bite you in the back when you weren’t looking. And we can definitely reduce the risk of that happening.
A year has gone by. The service evolved through this time, as we added more insights into it. Time to look at what we’ve doneWebRTC Insights by the numbers
We started small. The first WebRTC Insights issue looked at 6 issues, 7 PSAs and 2 market insights. 4 pages in total. Now we’re at 15-20 issues on average (twice as much when a Safari release happened) and 10 pages (or more).
In numbers, over the year this turned out to be:
26 Insights issues, 331 issues & bugs, 120 PSAs, 17 security vulnerabilities, 74 market insights and 185 pages. Phew…Bugs
In the past decade we have had more than 13,000 issues filed against libwebrtc, Google’s implementation of WebRTC that we all use in Chrome (and all other browsers in one way or another), with close to 5,000 of them external bug reports. In addition to that close to 2,000 external chromium bugs related to WebRTC.
WebRTC is a complex piece of software and staying on top of it requires quite some effort. While the development activity on WebRTC is much lower these days (at a third of the peak change rate back in 2017) there is still a surprising amount of issues we have to look at.
WebRTC Insights started from conversations about WebRTC issues and the challenges they bring between us. We have long looked at and discussed bugs, but this happened over chat and we never wrote it up. Nowadays we write up a summary, our thoughts and the potential impact each bug has. Quite often we learn something from it.
In the process we actually created an annotated list of issues that we can then refer to when we encounter new issues. So when Tsahi complained about an increase in video jitter statistics recently, Philipp just pointed him to the issue where we discussed this topic (you see, Tsahi’s memory isn’t what it used to be).Mailing lists and PSAs
“Public Service Announcements” or PSAs are a way for the WebRTC team (and Philipp) to communicate breaking changes in WebRTC. They range from changes to the C++ APIs to the plan-b deprecation and typically require action from developers using WebRTC in their applications.
Last but not least we do monitor the W3C working group and what happens there as it has a long term impact on where WebRTC is going.The crazy profession syndrome: WebRTC trials in Chrome
WebRTC uses field trials in Chrome to roll out changes that have some technical risk. We identify them which gives us insights into what might be a possible root cause for issues that are hard to reproduce locally. The best example for this recently was this report by Facebook where an experimental change to reduce the noise during opus dtx caused a large AV desync issue. We had been tracking the experiment for a couple of weeks at that point.Security patches in WebRTC
We keep track of WebRTC related CVEs in Chrome (17 in the last twelve months), determine whether they only affect Chromium or when they affect native WebRTC and need to be cherry-picked into forks of the native library.Where is the market headed?
This part is the bird’s eye view that we offer. The rest of the insights are the low level details developers need. Here, we look at the bigger picture of what WebRTC is and the market forces around it.
We bump into tweets, posts, LinkedIn messages and other articles out there – and when we feel they are relevant and important to your work, we mention them. And explain where we see this trend headed and what you should be aware of.
The market insights are designed and handpicked for the clients we serve in WebRTC Insights.We’re evolving
Over time, we’ve evolved the service.
Security and Chrome trials were added later on. We are now experimenting ourselves with short video explainers of each libwebrtc release (=once a month) and its implications to developers. We got some great feedback on it, so we’re likely to keep it as part of our format.
There are now also 3 different plans to the WebRTC Insights:
Want to join us for the ride this coming year?
To learn more, check us out at WebRTC Insights
You can leave us a message there to get a sample copy of one of our latest insights issue
What the announcements of Subspace and Cloudflare on their Managed WebRTC TURN services mean for the industry.
In the past couple of months we’ve seen two new entrants to the managed WebRTC TURN business. After stagnation for many years, this small market niche is becoming interesting. REALLY interesting.Table of contents
TURN servers are used in WebRTC in order to get your sessions connected if there’s no direct route available. I am not going to go into the technical part of it, but I’d say that without TURN servers, not all of your WebRTC sessions will get connected. You don’t need it for all sessions, but for some, you won’t be able to work without it. They are an essential component that has its own category in my WebRTC Developer Tools Landscape.
At the end of the day, TURN servers act as intermediaries by relaying the media between two points.
Roughly speaking, you have 3 alternatives in how you can get these set up:
In this article, I will be ignoring the “everything and the kitchen sink” approach. Not because it is bad, but because if you’re just interested in a managed WebRTC TURN, then you probably want to control a bit more of your destination (more on that later).Challenges of using open source coturn in production
Let’s start with the self hosting approach. The leading choice today is to take coturn, a popular open source TURN server, and deploy it on your own. There are one or two other alternatives, but this is by far the most common one.
The challenge though stems from the fact that for TURN the majority of the issues aren’t around integration or development but rather in configuration and maintenance. As such, it falls into the laps of ops, but requires knowledge and understanding of WebRTC.
The main culprit? The fact that you don’t need TURN for each and every session – and that there are 3 different TURN transport protocols, offering a progressive fallback mechanism.
What does that mean?
You install and configure your TURN server. But how do you test that all went well? Just conducting a WebRTC session will not tell you that. If the session succeeded, is it because it didn’t need TURN or because it used your TURN server properly? And if it did use it properly, was that on all 3 different transport protocols?
Configuring TURN is a headache:
In the early days of WebRTC, developers had two main alternatives:
You either knew what you were doing or didn’t want to know what you were doing.
The initial indication for managed WebRTC TURN service came from two vendors. It started with Xirsys and continued with Twilio.Xirsys
Xirsys was the first vendor to offer a managed WebRTC TURN service commercially. It was limited to a data center or two when they started, but grew over time.
Today, the Xirsys Cloud service spans 7 regional data centers.Twilio
Twilio is the most widely known CPaaS vendor out there. It is playing the best of suite game, with its large and growing portfolio of services. One of these products is their Twilio Global Network Traversal Service, a half-hidden product that enables you to leverage their TURN servers for your application without using their other CPaaS and WebRTC products.
At the time of writing, Twilio runs its media over 9 different regions, all on AWS.Why use a managed WebRTC TURN service?
I guess it is a matter of experience and expertise. Do you really want to deal with questions such as how do you decide which TURN server to connect a user to? How to deal with WebRTC TURN geolocation?
A managed WebRTC TURN service eventually targets the exact pain points and challenges that setting up your own TURN servers pose:
The best thing about managed WebRTC TURN services?
There’s no vendor lock-in.
Switching from one managed WebRTC TURN service to another or to your own self installed servers is a breeze – just change the iceServers configuration on your peer connections in WebRTC and you’re done. Theoretically, that’s a single line of code change.
It is also why I suggest anyone who is building their own WebRTC application to start by using a managed WebRTC TURN service – they can always switch to their own, and the cost of switching next year will be the same as just building it today. And as the lazy person that I am, I will always postpone to tomorrow something that I don’t have to do today.Managed WebRTC TURN – the post-pandemic version
Then came the pandemic, with its lockdowns, quarantine and the rise in use of WebRTC and any other remote communications technology.
The market stayed roughly the same for managed WebRTC TURN servers, or at least it did until 2021. What happened is that we now have 2 more vendors in this domain: Subspace and Cloudflare. And they are different: they are bigger in the physical footprint they have and they make use of Anycast – an IP addressing and routing scheme used to connect a large set of globally spread servers via a single IP address. This type of a solution also makes things a lot simpler to whitelist when needed.Subspace GlobalTURN
Subspace offers better connectivity than the open internet. They do that by optimizing the routes your packets go through. What you do is send your packets through their network, which will then figure out the best route.
In 2021, they decided to expand what they are doing to WebRTC as well, offering their GlobalTURN service. With around 100 cities and an Anycast addressing scheme, they offer a global footprint.
For Subspace, this isn’t the first VoIP related product they offer, but it is the first WebRTC related one. Would they move towards hosting media servers as well? I think it is an unlikely path for them.Cloudflare WebRTC Components
Cloudflare announced their own deployment of a managed WebRTC TURN service called WebRTC Components. Besides it being a TURN service, there’s not much to go by yet.
What we do know is that it relies on Cloudflare’s anycast network spanning 250+ cities.
For Cloudflare, this is the first WebRTC related offering, which was announced alongside a slew of other capabilities, targeted at cloud vendors (their R2 storage which directly competes with AWS S3 for example). There’s a good overview of the disruption path Cloudflare is taking. The WebRTC addition to it is an interesting choice.
Interestingly, I debated the potential of using Cloudflare’s Workers as a TURN service enabler when it was announced. Seems like they decided to build it on their ownWhich managed WebRTC TURN service to use?
That should be the question you should ask yourself.
It isn’t about whether you should use a managed WebRTC TURN service or deploy your own – it should be which managed WebRTC TURN service to select. Why? Because this is super simple to adopt and replace with zero vendor lock-in.
Pricing is important, but also global footprint, latency and quality. Then there are things like actually doing its job – the percentage of successful connections you get with it.
It will be interesting to see if and how Xirsys and Twilio address the threat from the newcomers to this market niche. For Xirsys this should be more worrying than it is for Twilio, as that’s one of their core products, whereas for Twilio it is a small part of what they offer to their customers.
Who would have thought that in 2021 we will see competition and innovation coming to the managed WebRTC TURN service?
One of WebRTC’s great features is its mandated strong encryption. Encryption mechanisms are built-in, meaning developers don’t (often) need to deal with the details. However, these easy, built-in encryption mechanisms assume you have: 1) media is communicated peer-to-peer and 2) a secure signaling channel setup. Most group-calling services make use of a media server device, […]
The post How does WebRTC End-to-End Encryption work? Matrix.org example (Dave Baker) appeared first on webrtcHacks.
Apple released iOS 15 with iCloud Private Relay broken for WebRTC - it still divulges your IP address. This post walks through why and how the WebRTC API's use your IP address information and how you can check what IP addresses are gathered.
Are free minutes and accounts in WebRTC video API worth the trouble? I think not. Don’t choose your CPaaS vendor based on their “free” tier.
I am finalizing my 10th edition of Choosing a WebRTC API report these days. In the past year I’ve heard from a few vendors and developers questions about the free tiers in this space. So I took the time as part of this edition, to sit down and analyze the price plans of the various vendors in the market and create another article as part of the report (one that is available through the membership site for those who purchase the report).
In this article, I want to shine a light on one aspect of price plans in WebRTC APIs which is the free tier.
Let’s dive into things, shall we?Table of contents
14 out of 24 vendors I looked at practice per minute pricing. Sometimes, they have multiple price strategies, but per minute pricing is the most common – especially on the bigger more widely known vendors.
Out of the 14 vendors, 5 offer free tiers in one way or another. And 2 offer credits – Amazon Chime SDK and Microsoft Azure Communication Services – these two offer IaaS cloud credits to startups as general practice and their CPaaS/WebRTC offering wraps into these as well (I’ve written about cloud giant effect on the CPaaS market last year).
Not all WebRTC API vendors offer a free tier
Free tiers seem to be almost “random” in who offers them and who doesn’tFree depends on the plan
Some vendors have free plans that depend on different things.
For Twilio, for example, free minutes come only with their Twilio Video WebRTC Go service, which… amounts to ~$10/month, and offers a limited peer-to-peer experience.
With some vendors, the free plan is actually a limited free evaluation for 1-4 months in timeframe.
That said, the most popular alternative seems to be free minutes on a paid plan. You give your credit card, and will only be charged if you pass a number of minutes on a given month. More on that – in the next section.
Free monthly minutes depend on the plan/feature set you choose/use
It might also be dependent on what you pay (did we say free plan?)10,000 free WebRTC minutes
Most vendors that give free minutes, are giving 10,000 free minutes per month.
Some give less. A few give more. The highest is 30,000 minutes per month.
If your service offers group calls of 10 participants for 30 minutes each time on average, then a single group call will take 300 minutes. That means ~33 such calls a month are free. Or a bit over a call a day.
This isn’t much. Not even for a small vendor just starting out. To be clear – this isn’t to say that 10,000 free minutes isn’t nice. Just that it won’t get you far.
The number of free minutes offered may seem a lot, but calculated for a use case they aren’t that many
Many small vendors see upwards of a million video minutes a month, so this amount to 1% of less of their total monthly minutes. Negligible in the long runWebRTC video free tier? Money Time
Minutes are nice, but how about money? How much money do you actually save with these free minutes?
I did the math. The numbers range between $30-$90 per month. Less than $1,000 per year.
If you are building a business and making your long term plans on the CPaaS vendor to use based on a potential discount of $1,000 a year then you’re doing it wrong.
Why aren’t CPaaS vendors offering higher free plans? Because they have costs they need to cover. Assuming a 10% cost over that price point, then 1,000 “free” accounts will cost them up to $100,000 a year to maintain. And that doesn’t include the support costs which are higher.
CPaaS vendors would like to have startups sample and use their service, but they also need to operate as a business and make money. Giving more minutes than they do today probably isn’t going to accomplish more paying customers – it will just bring in more free riders that will also leach on their soul and support resources.
Free WebRTC video CPaaS plans worth less than $100/month
When making your decision on choosing a vendor, ignore that plan in your own business plan
As a CPaaS vendor, decide if you want such a free tier and what type of customers it is going to attractHow do you choose a WebRTC CPaaS vendor?
The answer to this question is definitely NOT through their free tiers or minutes…
To some extent, the decision is made these days via pricing. It is why I’ve written in this round of my report to include a special article dedicated to pricing of WebRTC calls in CPaaS services. This includes the leading metrics these platforms use for their price plans as well as price ranges for each vendor. For this analysis, I’ve also added Zoom Video SDK as another reference point for pricing.
The report itself introduces a new CPaaS vendor and removes another vendor. It also sports a new features set structure, one that is geared towards the changes in requirements made due to the pandemic.
This report is used today by:
This month, until the report gets officially published, there’s a $500 discount. You can use coupon code API2021LAUNCH when you purchase the report.Learn more about my report
Hiring WebRTC developers? Here are some things you need to know and consider, since finding WebRTC experts for a job is challenging.
You’re growing. Obviously. And you have this huge, important, strategic, one of a kind, critical project. And it requires WebRTC. Only thing missing is developers. Or should I say skilled WebRTC developers.
How do you go about finding, hiring and retaining WebRTC developers?
I wrote a short post on LinkedIn the other day about this:
Typical conversation on #WebRTC recruitment
You: “Do you know any developer who can help us with WebRTC?”
Me: “No. Those I know either have a day job they love are are freelancers not looking for work (and almost always fully booked)”
You: “If you learn of a developer available let me know“
Me [Thinking ]: “Join the club at the end of that waiting list…”
Finding developers that know WebRTC is really hard. Seriously.
There’s a lot more demand than supply in this one, and the market is tiny compared to other technologies you need to deal with.
If you’re looking for WebRTC developers you can either:
poach someone from another vendor who does WebRTC. Tricky and expensive
find someone with the inclination and train him on WebRTC
If you’re on that second track of training, I can help you.
This brought with it a request to write this in longform so Philipp Hancke will have a place to refer recruiters to…
yes. Tsahi, please write a blog post so I can have a canned response for recruiters— Philipp Hancke (@HCornflower) August 5, 2021
Philipp – this one’s for youTable of contents
Oh – and if you are interested in history, this isn’t a new topic here. I wrote about finding WebRTC developers years ago…WebRTC developers: A supply problem
The chart above shows a crude comparison between WebRTC usage and LinkedIn profiles. While the pandemic has shown a huge increase in WebRTC usage (=demand) the change in LinkedIn profiles has been relatively moderate (=supply).
Here’s the two separate charts showing each data point independently:
LinkedIn profiles showing “WebRTC” in them grew steadily from ~17,000 to 25,000 profiles (47% growth in total) whereas growth in WebRTC usage (calculated as calls to GetUserMedia in page loads) grew 0.05 to 0.22 (340% growth in total), peaking at almost 0.6 with the pandemic (that’s %1,100 growth).
We’ve got a supply problem with WebRTC. There’s a shortage of developers, architects, product managers, testers and support who are savvy enough with WebRTC. They are all hard to come by, and it is harder still to know what they really know about WebRTC – installing your own Jitsi server and playing with it is different than running it at scale or developing your own SFU media server from scratch.
With this in mind, you can safely assume that one of the most popular topics raised when people talk to me about WebRTC is hiring WebRTC developers – or more accurately, if I can recommend anyone specific.The challenging skillsets of WebRTC
Why is it that it is hard to find WebRTC developers?
I think it starts from the diagram below:
WebRTC is multidisciplinary by its nature. It is located right between web and VoIP technologies:
This means a developer who needs to handle WebRTC needs to have a good grasp of more than a single field of software development. And this isn’t easy to come by.
There’s one more reason though, and that’s the fact that WebRTC means different things to different people, and isn’t really focused on a single set of skills. Look at the short set of questions I’ve asked years ago about how much WebRTC developers are worth. The answers are mostly around “it depends”, where it depends on what tasks or job description that developer is filling up.
Here are the main areas today that you may need to find different profiles of WebRTC developers:
In each domain, the skillset is slightly different and you will be hard pressed to find a superhero developer that meets all your requirements in all areas.Hiring WebRTC talent
WebRTC hiring is challenging. If you are looking for talented engineers who know a thing or two about WebRTC, then you are in for a world of pain. Finding them isn’t easy and hiring them is even harder.
Here are the different techniques I’ve seen vendors take when trying to find and hire WebRTC engineers.WebRTC head-hunting and poaching
You can go head hunting for WebRTC talent. Bear in mind 3 things though:
There are two approaches here. Let’s call them bottom up and top down.
Bottom up – you find the individual developers that fit the profile you are looking for, and then you reach out to them to see if they are bored enough to consider moving elsewhere
Top down – target a vendor in this space who you think peaked or someone who got acquired or just someone you think a bit vulnerable and attractive as an employer, and then figure out who are the developers there worth approaching to poach
Neither approach is easy. They are time consuming, frustrating and long.Job boards and job listings
You could use traditional job boards and job listing sites, place the job opening on your website, etc. What you’ll most probably get is going to be generalists with little domain knowledge and expertise in WebRTC. This means most applicants won’t have the WebRTC experience you seek.
The only other option here is to do an ad placement on WebRTC Weekly and/or webrtcHacks – many of the sponsors there use it for job listings, and you can try as well. The main advantage here is that the readership is quite relevant – developers working with WebRTC.
* Note that I operate WebRTC Weekly and affiliated with webrtcHacksHire from an adjacency
This is something I suggest to many of my clients. Hire from an adjacency:
My favorite is probably finding companies that vanished, for example Polycom Israel. They had a large engineering team in Israel experienced in video conferencing. You can try to find developers who worked there 5-10 years ago and… moved on – often to other domains. And try to get them back. They won’t be experts in WebRTC, but they’ll know a lot about how to handle real time video. And that’s better than nothing.
The same is applicable elsewhere in the world and in other adjacencies.
When hiring from an adjacency though, you will need to be certain the candidate in question isn’t “in love” in how things are done today and have the willingness and the openness to learn and grow. WebRTC brings with it new paradigms and challenges and developers who have partial experience and knowledge from an adjacency need to be open to learn new concepts.Nurture and grow in-house WebRTC expertise
When all else fails, you’ll need to grow someone in-house or train a new hire that is clueless about WebRTC to become that expert. Not an easy task, but certainly achievable.
WebRTC requires a certain inclination. There’s a need to wrap your head around asynchronous events and programming (lots of await and callbacks). There’s a need to understand codecs and lossy compression mechanisms (at least at the conceptual level). There’s perpetual optimization and fine tuning work that goes with it. Not everyone likes to work in such environments (I thrive in them).
Once you find that person, you will need to train him. Something that again can happen in one of 3 ways:
I have only discussed developers so far, but the product life-cycle of WebRTC products involves more than just the engineers who need to understand WebRTC. There are a few more roles to think about:
All of these roles need a solid understanding of WebRTC if it is part of the things you are offering in your company.Can I help?
There are several things that I actively do here:
If you’re interested in learning more, feel free to contact me.
Oh – and don’t ask me if I know someone suitable. You’re likely not the first to ask me that this week.
Philipp Hancke discusses a how to properly release Media Element resources with WebRTC and a recent Chrome issue that apps to stop handling larger numbers of participants.
The post Dealing with HTMLMediaElements and srcObjects in WebRTC applications appeared first on webrtcHacks.
WebRTC video quality requires some tweaking to get done properly. Lets see what levels we have in the form of bitrate, resolution and frame rate available to us.
Real time video is tough. WebRTC might make things a bit easier, but there are things you still need to take care of. Especially if what you’re aiming for is to squeeze every possible ounce of WebRTC video quality for your application to improve the user’s experience.
This time, I want to cover what levers we have at our disposal that affect video quality – and how to use them properly.Table of contents
Video plays a big role in communication these days. A video call/session/meeting is going to heavily rely on the video quality. Obviously…
But what is it then that affects the video quality? Lets try and group them into 3 main buckets: out of our control, service related and device related. This will enable us to focus on what we can control and where we should put our effort.Out of our control From my workshop on WebRTC innovation and differentiation
There are things that are out of our control. We have the ability to affect them, but only a bit and only up to a point. To look at the extreme, if the user is sitting in Antarctica, inside an elevator, in the basement level somewhere, with no Internet connection and no cellular reception – in all likelihood, even if he complains that calls aren’t get connected – there’s nothing anyone will be able to do about it besides suggesting he moves himself closer to the Wifi access point.
The main two things we can’t really control? Bandwidth and the transport protocol that will be used.
We can’t control the user’s device and its capabilities either, but most of the time, people tend to understand this.Bandwidth
Bandwidth is how much data can we send or receive over the network. The higher this value is, the better.
The thing is, we have little to no control over it:
None of this is in our control.
And while we can do minor things to improve this, such as positioning our servers as close as possible to the users, there’s not much else.
Our role with bandwidth is to as accurately as possible estimate it. WebRTC has mechanisms for bandwidth estimation. Why is this important? If we know how much bandwidth is available to us, we can try to make better use of it –
Over-estimating bandwidth means we might end up sending more than the network can handle, which in turn is going to cause congestion (=bad)
Under-estimating bandwidth means we will be sending out less data than we could have, which will end up reducing the media quality we could have provided to the users (=bad)Transport protocol
I’ve already voiced my opinion about using TCP for WebRTC media and why this isn’t a good idea.
The thing is, you don’t really control what gets selected. For the most part, this is how the distribution of your sessions is going to look like:From my Advanced WebRTC Architecture Course
Why is that? Just because networks are configured differently. And you have no control over it.
You can and should make sure the chart looks somewhat like this one. 90% of the sessions done over TURN/TCP should definitely raise a few red flags for you.
But once you reach a distribution similar to the above, or once you know how to explain what you’re seeing when it comes to the distribution of sessions, then there’s not much else for you to optimize.Service related
Service related are things that are within our control and are handled in our infrastructure usually.This is where differentiation based on how we decided to architect and deploy our backend will come into play.Bitrate
While bandwidth isn’t something we can control, bitrate is. Where bandwidth is the upper limit of what the network can send or receive, bitrate is what we actually send and receive over the network.
We can’t send more than what the bandwidth allows, and we might not always want to send the maximum bitrate that we can either.
Our role here is to pick the bitrate that is most suitable for our needs. What does that mean to me?
It is important to remember to understand that increasing bitrate doesn’t always increase quality. It can cause detrimental decreases in quality as well.
Here are a few examples:
There are a lot of other such cases as well.
So what do we do? I know, I am repeating myself, but this is critical –
Codecs affect media quality.
With video, this is a lot more nuanced. You have a selection of VP8, VP9, H.264, HEVC and AV1.
Here are a few things to consider when selecting a video codec for your WebRTC application:
Choosing a video codec for your service isn’t a simple task. If you don’t know what you’re doing, just stick with VP8 or H.264. Experimenting with codecs is a great time waster unless you know your way with them.Latency How you design your WebRTC infrastructure will affect the latency
While we don’t control where users are – we definitely control where our servers are located. Which means that we can place the servers closer to the users, which in turn can reduce the latency (among other things).
Here are some things to consider here:
Measure the latency of your sessions (through rtt). Try to reduce it for your users as much as possible. And assume this is an ongoing never-ending process
Here’s a session from Kranky Geek discussing latencies and media servers:Looking at scale and servers
There’s a lot to be said about the infrastructure side in WebRTC. I tried to place these insights in an ebook that is relevant today more than ever – Best practices in scaling WebRTC deploymentsDevice related
You don’t get to choose the device your users are going to use to join their meetings. But you do control how your application is going to behave on these devices.
There are several things to keep in mind here that are going to improve the media quality for your users if done right on their device.Available CPU
This should be your top priority. To understand how much CPU is being used on the user’s device and deciding when you’ve gone too far.
What happens when the device is “out of CPU”?
So what did we have here?
You end up with poor video quality and video freezes
The network gets more congested due to frequent requests for I-frames
Your device heats up and battery life suffers
Your role here is to monitor and make sure CPU use isn’t too high, and if it is, reduce it. Your best tool for reducing CPU use is by reducing the bitrates you’re either sending and/or receiving.
Sadly, monitoring the CPU directly is impossible in the browser itself and you’ll need to find out other means of figuring out the state of the CPU.Content type
With video, content and placement matter.
Let’s say you have 1,000kbps of “budget” to spend. That’s because the bandwidth estimator gives you that amount and you know/assume the CPU of both the sender and receiver(s) can handle that bitrate.
How do you spend that budget?
WebRTC makes its own decisions. These are based on the bitrate available. It will automatically decide to increase or reduce resolution and frame rate to accommodate for what it feels is the best quality. You can even pass hints on your content type – do you value motion over sharpness or vice versa.
There are things that WebRTC doesn’t know on its own through:
It is going to be your job to figure out these things and place/remove certain restrictions of what you want from your video.Optimizing large group calls
The bigger the meeting the more challenging and optimized your code will need to be in order to support it. WebRTC gives you a lot of powerful tools to scale a meeting, but it leaves a lot to you to figure out. This ebook will reveal these tools to you and enable you to increase your meeting sizes – Optimizing Group Video Calling in WebRTCThe 3-legged stool of WebRTC video quality
Video quality in WebRTC is like a 3-legged stool. With all things considered equal, you can tweak the bitrate, frame rate and resolution. At least that’s what you have at your disposal dynamically in real-time when you are in the middle of a session and need to make a decision.
Bitrate can be seen as the most important leg of the stool (more on that below).
The other two, frame rate and resolution are quite dependent on one another. A change in one will immediately force a change in the other if we wish to keep the image quality. Increasing or decreasing the bitrate can cause a change in both frame rate and resolution.Follow the bitrate
I see a lot of developers start tweaking frame rates or resolutions. While this is admirable and even reasonable at times, it is the wrong starting point.
What you should be doing is follow the bitrate in WebRTC. Start by figuring out and truly understanding how much bitrate you have in your budget. Then decide how to allocate that bitrate based on your constraints:
Always start with bitrate.
Then figure out the constraints you have on resolution and frame rate based on CPU, devices, screen resolution, content type, … and in general on the context of your session.
The rest (resolution and frame rate) should follow.
And in most cases, it will be preferable to “hint” WebRTC on the type of content you have and let WebRTC figure out what it should be doing. It is rather good at that, otherwise, what would be the point of using it in the first place?Making a choice between resolution and frame rate
Once we have the bitrate nailed down – should you go for a higher resolution or a higher frame rate?
Here are a few guidelines for you to use:
I’ve had my fair share of discussions lately with vendors who were working with WebRTC but didn’t have enough of an understanding of WebRTC. Often the results aren’t satisfactory, falling short with what is considered good media quality these days. All because of wrong assumptions or bad optimizations that backfired.
If you are planning to use WebRTC or even using WebRTC, then you should get to know it better. Understand how it works and make sure you’re using it properly. You can achieve that by enrolling in my WebRTC training courses for developers.Learn more about my WebRTC training
The post Tweaking WebRTC video quality: unpacking bitrate, resolution and frame rates appeared first on BlogGeek.me.
Introduction to capture handle - a new Chrome Origin Trial that lets a WebRTC screen sharing application communicate with the tab it is capturing. Examples use case discussed include detecting self-capture, improving the use of collaboration apps that are screen shared, and optimizing stream parameters of the captured content.
The post Identifying Shared Tabs using Capture Handle (Elad Alon) appeared first on webrtcHacks.
How to seperate multiple people in the same camera feed into their own unique video streams that can be individually transmitted Google's MediaPipe and the W3C's new MediaStreamTrack API
The post Making Zoom’s Smart Gallery on the Web with MediaPipe and BreakoutBox appeared first on webrtcHacks.
When using WebRTC you should always strive to send media over UDP instead of TCP. at least if you care about media quality
Every once in a while I bump into a person (or a company) that for some unknown reason made a decision to use TCP for its WebRTC sessions. By that I mean prioritizing TURN/TCP or ICE-TCP connections over everything else – many times even barring or ignoring the existence of UDP. The ensuing conversation is usually long and arduous – and not always productive I am afraid.
So I decided to write this article, to explain why for the most part, WebRTC over UDP is far superior to WebRTC over TCP.Table of contents
Since the dawn of time the internet, we had UDP and TCP as the underlying transport protocols that carry data across the network. While there are other transports, these are by far the most common ones.
And they are different from one another in every way.
UDP is the minimal must that a transport protocol can offer (you can get lower than that, but what would be the point?).
With UDP you get the ability to send data packets from one point to another over the network. There are no guarantees whatsoever:
No guarantees. Did I mention that part?
With TCP you get the ability to send a stream of data from one point to another over a “connection”. And it comes with everything:
That guaranteed delivery requires the concept of retransmissions – what gets lost along the way needs to be retransmitted. More on that fact later on.
We end up with two extremes of the same continuum. But we need to choose one or the other.TCP rules the web
Reading this page? You’re doin’ that over HTTPS.
HTTPS runs over a TLS connection (I know, there’s HTTP/3 but bear with me here).
And TLS is just TCP with security.
And if you are using a WebSocket instead, then that’s also TCP (or TLS if it is a secure WebSocket).
No escaping that fact, at least not until HTTP/3 becomes common place (which is slightly different than running on top of TCP, but that’s for another article).
Up until WebRTC came to our lives, everything you did inside a web browser was based on TCP in one way or another.UDP rules VoIP
VoIP or Voice over IP or Video over IP or Real Time Communications (RTC) or… well… WebRTC – that takes place over UDP.
Why? Because this whole thing around guaranteed delivery isn’t good for the health of something that needs to be real time.
Let’s assume a latency of 50 milliseconds in each direction over the network, which is rather good. This translates to a round trip time of 100 milliseconds.
If a packet is lost, then it will take us at least a 100 milliseconds until the one who sent that packet will know about that – anything lower than that won’t allow the receiver to complain. Usually, it will take a bit more than 100 milliseconds.
For VoIP, we are looking to lower the latency. Otherwise, the call will sound unnatural – people will overtalk each other (happens from time to time in long distance calls for example). Which means we can’t really wait for these retransmissions to take place.
Which is why VoIP, in general, and WebRTC in particular, chose to use UDP to send its media streams. The concept here is that waiting will cause a delay for the whole duration of the session reducing the experience altogether, while the need to deal with lost packets, trying to conceal that fact would cause minor issues for the most part.
With WebRTC, you want and PREFER to use UDP for media traffic over TCP or TLS.WebRTC ICE: Preferences and best effort
We don’t always get what we want. Which is why sometimes our sessions won’t open with WebRTC over UDP. Not because we don’t want them to. But because they can’t. Something is blocking that alternative from us.
That something is called a firewall. One with nasty rules that… well… don’t allow UDP traffic. The reasons for that are varied:
This means that you’ll be needing TCP or TLS to be able to connect your users on that WebRTC session.
But – and that’s a big BUT. You don’t always want to use TCP or TLS. Just when it is necessary. Which brings us to ICE.
ICE is a procedure that enables WebRTC to negotiate the best way to connect a session by conducting connectivity checks.
In broad strokes, we will be using this type of logic (or strive to do so):
The diagram above shows the type of preferences we’d have while negotiating a session with ICE.
UDP comes first.When is TCP (or TLS) good for WebRTC media?
The one and only reason to use TCP or TLS as your transport for WebRTC media is because UDP isn’t available.
There. Is. No. Other. Reason. Whatsoever.
And yes. It deserved a whole section of its own here so you don’t miss it.
TCP for me is a last resort for WebRTC. When all else failsWhen will TCP break as a media transport for WebRTC?
The moment you’ll have packet loss on the network, TCP will break. By breaking I don’t mean the connection will be lost, but the media quality you’ll experience will degrade a lot farther than what it would with UDP.
Packet loss due to congestion is going to be the worst. Why? Because it occurs due to a switch or router along the route of your data getting clogged and starting to throw packets it needs to handle.
Here are all the things that will go wrong at such a point:
Bottom line – TCP causes packet loss issues to worsen the situation a lot further than they are, with a lot less leeway on how to solve them than we have running on top of UDP.
The assumptions TCP makes over the data being sent are all wrong for real time communications requirements that we have in protocols like WebRTCTime to learn WebRTC
I’ve had my fair share of discussions lately with vendors who were working with WebRTC but didn’t have enough of an understanding of WebRTC. Often that ends up badly – with solutions that don’t work at all or seem to work until they hit the realities of real networks, real users and real devices.
I just completed a massive update to my Advanced WebRTC Architecture training course for developers. In this round, I also introduced a new lesson about bandwidth estimation in WebRTC.
Next week, we will start another round of office hours as part of the course, letting those taking this WebRTC training ask questions openly as well as join live lessons on top of all the recorded and written materials found in the course.
If you are planning to use WebRTC or even using WebRTC, there isn’t going to be any better timing to join than this week.Learn more about my WebRTC training
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It seems like CPaaS vendors have grown complacent compared to the rapid innovation coming from UCaaS vendors. This makes no sense.
CPaaS has been leading the innovation when it comes to how developers build communication products. This has been the case ever since CPaaS was coined. But now, the trend is changing. This is doubly true for WebRTC and video communication services. UCaaS vendors have taken the lead in innovation and setting the pace of the market, leaving CPaaS vendors behind.
Can this trend be reversed? Is this a bad omen for CPaaS vendors competing in video use cases?Table of contents
I used to work at RADVISION. The company specialized in video conferencing equipment but was split into two business units. The one I was a part of licensed VoIP software stacks to developers. You could say that what we did predates CPaaS. We didn’t have the cloud or server APIs but we sure did have SDKs.
In each and every townhall the company had, the CEO used to mention that our business unit was a precursor of the industry. Whatever requirements we’ve seen, whatever trend we experienced in sales (increase or decrease) was just an indicator of what is to come in the market in 3 years or so. The reasoning was simple – we licensed to developers, which then built their products and put them to market. Development cycles being as they were, 3 years was a good estimate.
Fast forward to today, and you have CPaaS vendors (the technology licensors of communication development tools) and the rest of the industry. And the large part of the rest of the industry is UCaaS.
The thing is, UCaaS vendors are no longer waiting for CPaaS vendors to innovate – they are just doing it on their own.The promise of CPaaS
Communication Platform as a Service. What is it for anyways?
The whole purpose of CPaaS is to reduce the time to market for developers. Make it easier to get things done with communications by developing all the nasty little details for you.
Call it low code. Call it SDK or API or whatever.
I did an interview with Jeff Lawson, CEO of Twilio years ago. There Jeff explains the essence of Twilio – why he started the company. And the reason is to solve the communication problem for companies so they can focus on building great customer experiences.
Remember this one. We will be back to this interview a wee bit later.Pandemic requirement shifts
Then the pandemic hit. And with it, a change in what communication requirements looked like around the world for all use cases.
4 distinct changes took place:#1 – meetings became larger
We had large meetings before. The difference was that we connected rooms with groups of people in each room. Now? Everyone’s joining from his own place.
A meeting with 20 people in 3 rooms became a meeting with 20 people from 20 rooms. We will be back in the office, but the requirement for bigger meetings, with more people joining remotely will still be there with us.
Look at the start of this session from last year’s Kranky Geek virtual event.
Here Li-Tal Mashiach, Senior Engineering Manager at Facebook in the Messenger team explains what they’ve seen as changes in the usage of video calls in Messenger. Look at around the 2:40 mark in that video.#2 – more meetings for longer periods of time
This one is obvious. Or is it?
Almost all vendors have seen a significant growth in both the number of video sessions conducted on their platforms as well the length of these sessions.
Scale had to be dealt with across these two axes.
You need to make sure you can carry conversations that now take hours on end instead of minutes:
In both cases, much of the interaction is just ambient video. They do things together or apart and just have these social interactions take place because they can’t meet. Funny enough, my son and his cousin aren’t stopping it now even though everything is open – that’s because meeting physically requires a 20 minute car ride…
How does that change the focus? How do you maintain servers, upgrade and update them when sessions can take hours on end on a machine? Does it mean the media servers also need to be stabler in how they operate?
And what about the number of sessions? Is it that easy to scale 10x or more your current traffic? This isn’t a simple question to contend with. Google shared their own challenges with scaling Meet which makes for a fascinating read. I had my share of vendors to help with best practices in scaling their WebRTC infrastructure during the last 15 months as well.#3 – more networks
Back to that Kranky Geek video by Facebook. They saw an increase in desktop access. More than they had expected being mobile first.
I’d argue that we’ve all seen more variety in devices and networks. My apartment went from 1 video calling user to 4 video calling users in a matter of a day. Billion people or more who never went on a video call have done so and will continue to do so at least some of the time.
What devices do these billion people have? What does their home network look like?
If you look at the technology adoption curve, these aren’t the innovators or early adopters. They aren’t even the early majority. They include both the late majority and the laggards.
This means we’re facing a lot more variance in devices and networks. In the need to deal with lower end capabilities and resources available. And to deal with having these large groups take place with a larger variety of the differences across devices.#4 – more places
The best part of video calling during the lockdowns and up until today is taking a peak at other people’s home office. You get to see a piece of who they really are outside “work”.
These places are almost always less than ideal.
Everything you can think of that affects the audio and video quality due to external sources will be there. And you can’t always ask the user to go purchase a better camera, change where he is sitting or replace his device.
It becomes a technical problem to solve many of these issues, especially when the service offers ad-hoc connectivity for its users.CPaaS during the pandemic
CPaaS were supposed to help vendors build their products. Look at future needs and cater for them. And for the most part they do. But somehow during this pandemic, it seems that many of them have failed to do so.
I’ll look at Twilio here – and not because they are the only vendor with these issues – but because they are the biggest CPaaS vendor and the precursor of the industry.
Last year after Twilio’s Signal 2020 event I wrote that I expected more of them:
For me this says that Twilio hasn’t invested in video as much in the last year or two. If they had, they would have announced something more thrilling and interesting. Maybe larger meetings, above 50 participants? Broadcasting capabilities? Noise suppression? Something…
Since I wrote that, 8 months have passed. Meeting sizes for Twilio Programmable Video are still limited to 50 participants. There are no broadcasting capabilities. No noise suppression. No background blurring. Nothing.
I can’t even recall any real additional feature that Twilio introduced for Twilio Programmable Video since that Signal event. Maybe updates and improvements to their React reference app, but nothing more.
Most other vendors showed similar inclination and introduction of new features throughout the pandemic. It seems like the trend now for video APIs is to focus on embedded iframes for faster development. These have been discussed and experimented with years ago, and now seem to be finding new traction and interest.
It takes more time to develop features in CPaaS than it does on other platforms. The reason for that is the CPaaS vendors need to do 2 things others don’t have to deal with:
But let’s face it. These new requirements have been around for 15 months now…
There are obviously a few caveats here:
I am griping here about video
CPaaS has grown during the pandemic, so this hasn’t hurt them. Yet
Video is usually a small percentage of traffic and income for a CPaaS vendorUCaaS during the pandemic
UCaaS shows a stark contrast to how CPaaS responded.
Many of the leading vendors have added background blurring and replacement, noise suppression and other features and capabilities. They have done so in breakneck speeds and they seem to be spewing out new features every week or so.
This isn’t limited to a single vendor. Out of the top of my head: Zoom, Microsoft Teams, WebEx, Google Meet and RingCentral all introduced these features in the past year. And all of them seem to be investing further into these areas while pushing forward other initiatives they have, each with its own focus.
Remember Jeff’s interview? I asked him if he believed UC vendors should develop their services on top of CPaaS. This is what he answered:
Yeah. I believe that companies whose primary business is communications can and definitely should and would get competitive advantage by using a platform like Twilio to build upon. The reason why is this. It used to be when those UC companies started, their core competency was making the phone ring. Then they’d add some software functionality on top of it, sure, but the vast majority of what they worried about was how do I make the phone ring? The problem is Twilio has democratized that ability.
The existing UCaaS vendors, they would be wise to build on top of the same platform that any developer in the world can come and start to compete with them on. If they don’t, those independent software developers, they can actually start and build companies that are really compelling competitors, because they don’t have to focus on the low level bits. They’re focused on the things customers really care about, which is features, functionality, and the user experience that matters.
While mostly true, this doesn’t hold water these days for video communications. Relying on CPaaS vendors means you need to figure out the feature set that is necessary to be a compelling competitor yourself – larger groups, background replacement, noise suppression, …
CPaaS vendors need to put their act together in the video domain, or start losing customers that will just go build this on their own. Especially when we see Zoom coming up with their Video SDK and becoming a direct competitor to CPaaS vendors.
UCaaS vendors are having their own headaches in the market due to the dramatic changes that Microsoft and Google are bringing into this domain. I’ll leave that for a future article.Pandemic valuations
The pandemic also changed the dynamics in communication vendor valuations, shifting the focus to slightly different domains.
Hopin and Clubhouse, which I already touched on in my previous article about the new era in WebRTC.
Agora (video CPaaS vendor) had a hugely successful IPO, followed by another spike due to the popularity of Clubhouse (who is using them). They are now back to roughly their initial IPO price point.
Twilio (CPaaS) increased in their valuation throughout the pandemic. My guess is that this is mostly due to the increased use in voice and SMS. Less so in video, where they invest a lot less.
Zoom. Need I say more?The differentiation dilemma & Build vs Buy
How does one differentiate then?
I am seeing the following trends in CPaaS adoption and use. They used to be related to pricing, but now they are becoming more and more related to feature sets and differentiation needs:
Most enterprises stick with the use of CPaaS vendors. They rely on them for their communication needs. They will switch from a CPaaS vendor to another CPaaS vendor if they can get better pricing or if their current vendor is lacking features (or provides poor support).
Technology vendors and startups will pick either CPaaS vendors as their starting point or prefer going it alone from the get go. Those that become hugely successful will end up actively working on replacing the CPaaS vendor with their own infrastructure. They will see that as an imperative a lot more than their enterprise brethrens.
Unified communication vendors will continue as they are. Assuming that communication infrastructure is core to their business and will work towards maintaining their own knowledge and experience in the area – doubly so after the pandemic.Wake up and smell the coffee
CPaaS vendors should wake up and smell the coffee.
The world has changed. Drastically.
There’s no going back to the old ways – even without quarantines.
I believe that there’s a competitive advantage waiting here. CPaaS vendors have been shying away from these requirements. The first ones to come out with actual solutions and feature capabilities that will ease the development of customers will win due to this differentiation.
The reason this hasn’t happened so far is that traditionally, such things weren’t catered for directly by CPaaS vendors – it is out of their comfort zone. This leads to an opportunity that is up for the taking.
On a similar note, after running successfully the Future of Communications workshop with Dean Bubley, we decided that it is both information packed and fun to do. If you are interested in a private session for your company – let us know.
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After 10 years, we are at the dawn of a new era for WebRTC. This one is going to focus on differentiation and will bring with it new dominant players into the field.
There’s a change in the air. I think it started towards the end of 2019, but now it is quite obvious to see. WebRTC is changing – not the specification but rather who is using it and how it is used.Table of contents
There’s a slide I showed last week in the workshop of the future of video and real-time communications. It resonated with me with the latest news of Justin Uberti leaving Google. So much so, that I decided to record it separately and share it here as well:
We’ve moved from exploration to growth and now into differentiation when it comes to WebRTC.
What got us there exactly?WebRTC 1.0
We’ve got that WebRTC 1.0 milestone behind us now.
I haven’t written any special article about WebRTC 1.0, because the main question you need to ask yourself is what changed?
And the real answer is nothing.
The work towards WebRTC 1.0 was important and this is an important milestone. But browser vendors already implemented WebRTC. And vendors already used WebRTC in browsers and native applications as if this was a done deal already.
If you were using WebRTC before, then nothing has changed for you since the announcement of WebRTC 1.0.
And if you haven’t used WebRTC yet, then why start now? What was holding you back so long? The fact that you weren’t sure if it is here to stay???
Having WebRTC 1.0 out is an important milestone. More a symbol and a signpost than anything else.The pandemic The pandemic had a positive effect on WebRTC adoption
The pandemic got us all quarantined and changed everything.
There’s no new normal to talk about yet, but if you’re believing things are “going back to normal” then you’re wrong.
To put simply:
These changes are bringing with them a lot of new demand, new use cases and new requirements.
What we focused on with WebRTC up until 2020 was suitable for the “old” pre-pandemic world. What we need to focus on now is on the “new” post-pandemic world, one which has slightly different requirements.Zoom Is Zoom the exception to prove the rule?
Even before the pandemic, Zoom’s IPO has been phenomenal.
After the pandemic, Zoom has become a household name.
Pick any communication service you wish from any vendor in the globe. Randomly pick 100 people from the world’s population. How many of them will know that vendor or service, and how many of them will know Zoom?
Zoom doesn’t really use WebRTC, so why should you?
This is an important question. The appropriate answer is probably one of context. Your context is different from Zoom’s.
And yet the WebRTC industry, its stack, the browsers and vendors are consistently being compared to Zoom.
Your ability to compete with Zoom on quality and connectivity is greatly dependent on Google, and what they decide to do with WebRTC.
You are not in full control over your destiny.WebRTC musical chairs
There were a few changes in the people who are working and dealing with WebRTC directly recently. I want to discuss 3 specific cases that I think mark the end of an era.Dr Alex Gouaillard, CoSMo and Millicast
Alex has been a known figure in the WebRTC community. His voice on subjects, his passion and his work has made its mark on our industry. He will be sorely missed.
In recent years, Alex focused heavily in the area of live streaming, trying to solve the challenge of broadcasting a WebRTC stream to many participants. He has been a vocal proponent of the use of AV1.
It will be interesting to see who will pick the mantle here and fill the void in explaining and promoting these use cases now.Nils Ohlmeier, Mozilla (now 8×8)
Nils Ohlmeier has been “the guy” from Mozilla who represented WebRTC in Firefox.
He shared the work Mozilla is doing in Firefox for WebRTC in last year’s virtual Kranky Geek event as part of the browsers panel we did:
Nils switched employers this month, starting to work at 8×8 in the role of Principal Engineer. He will be contributing to the Jitsi codebase and its growth. While Jitsi has a large and vibrant ecosystem, is it anywhere near the size and complexity Mozilla had to deal with?
Who is going to take this role at Mozilla?
Is Firefox interesting as a browser for WebRTC developers and users anymore?
Was it time to move on now that the biggest challenges of WebRTC for browser vendors is “behind” us?
To me, these questions more than anything else mark the change in times.Justin Uberti, Google Stadia (now Clubhouse)
Justin Uberti was there from the start when it came to WebRTC.
He is considered by many the lead engineer behind the Google Chrome team of WebRTC, and he was part of the original duo (not only the app) – Serge Lachapelle & Justin Uberti.
Justin moved on from the WebRTC team to Google Stadia at the end of 2019. He worked on Stadia related features before that as well.
This month, he decided to move on, leaving Google altogether, pursuing new activities. Justin is staying in the WebRTC industry, as his new role is Head of Streaming Technology at Clubhouse.
Here’s what Justin had to say at Kranky Geek 2018 during Google’s WebRTC update session:
It is truer today than it was in 2018…
Definitely the end of an era.WebRTC “winners” of 2021
In 2017 I’ve written about 10 Massive Applications Using WebRTC.
That was 3.5 years ago and before anyone thought about quarantines or Zoom.
Fast forward to today, and that list is going to look different.
Two vendors I want to highlight here are Hopin and Clubhouse. They are different from the other vendors we’ve seen in the past who are making use of WebRTC.Hopin
Hopin is a virtual events platform founded in June 2019, a bit less than two years ago. They couldn’t ask for a better timing (maybe start 6 months earlier?).
Within that timespan, Hopin managed to raise a whopping $571.4M in total and made 4 acquisitions (including StreamYard, Streamable and Jamm).
There are many virtual events platforms ever since the pandemic started but Hopin seems to be the biggest and most widely known one. They have shown that they aren’t shy of acquiring the technologies they need in order to get their feature set where they want it to be.
In 2019, who would have thought a virtual events vendor would be worth $5.65B in valuation by 2021?
Hopin has a nice warchest that they can use to grow their business, attract top notch developers and acquire or acquihire their way to success.Clubhouse
Another interesting vendor is definitely Clubhouse.
Everyone wants to be Clubhouse these days, but there’s still only a single Clubhouse out there.
Clubhouse started life with the pandemic, in March 2020. After only 14 months it has a valuation of $4B and has been funded well over $100M ($110M by series B in January this year, and another undisclosed series C in April). That’s quite a feat for a voice only, iOS only (until recently) service.
It has a warchest to rival that of Hopin and the same kind of hype behind it to allow it to do practically anything it wanted.
Clubhouse still lacks a real business case, but it doesn’t seem to be stopping it.
Clubhouse is known to be using Agora as their CPaaS vendor, but that may soon change. They hired Justin Uberti from Google, and the only reason for that to me seems to be the desire to own and control their infrastructure.Google
Google is still the big winner of WebRTC.
If you look at what features are added to WebRTC, then the answer to that is whatever Google needs for its own uses.
These uses now include Google Meet, Google Stadia and Google Assistant.
If your use case has the same requirements in general then you’re in good shape. If you are going “off the reservation”, then prepare for a life of misery if there’s something missing that you need and isn’t in Google’s own set of requirements.
WebRTC is open source up to a point. Not because the code isn’t open and available to all, but because the main implementation is owned and controlled by Google and the main browser you’ll need to work with is Chrome.Welcome to the new WebRTC
From now on, WebRTC is going to be different.
Talking heads are still an important part of it, but the focus is shifting from a “video chat” or “video conferencing” service into a communication service that is unique. What that is exactly is hard to say, but suffice to say that WebRTC is there in the background.
And fading to the background is exactly what we wanted from WebRTC – the technology is only great once we start forgetting it is there.
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How to use the AWS API Gateway WebSocket API functionality with Lamdba functions to implement a serverless WebRTC signaling architecture
The post How to Leverage the AWS WebSocket API for Serverless WebRTC signaling appeared first on webrtcHacks.
Rethink the way we look at interoperability and standardization in communications, now that we live in a WebRTC & Zoom world.
We live in a different world. This video popped up in my Facebook as something I shared a year ago:
It is in Hebrew, but there are enough words in English there to make it quite apparent. This comedian is trying to explain to his mother over the phone how to use Zoom.
Today? Everyone knows how to use Zoom. Or WebRTC.The transformation of communication technology
I started my professional adult life in a video conferencing company. There, I lived and breathed interoperability and standards for the better part of 13 years. I have a few contributions that got approved at the ITU and 3GPP. I’ve been to interoperability events and even hosted two of those in Israel.
At the time, the mindset was a telephony one:
There were two main reasons why we wanted such a world to live in:
Then Apple came with the iPhone and changed. Everything.
From embedded platforms, the smartphones became open programming platforms (open even within the closed gardens of their app stores).
Today, many of the embedded devices include an Android operating system, making it ever easier to develop software for them.
This brought with it a new kind of openness:
This didn’t mean standards were unimportant. It meant that interoperability became less interesting. Vendors could now bake their own proprietary additions on top of the standards that give extra features without the need to think too much about interoperability with other vendors – that’s because your client now brings his own device (BYOD) and you supply the software application to connect to the infrastructure.
Oh, and by the way – that infrastructure? It is now built in the cloud. And the cloud enables rapid development and hyper growth. Which again means that caring about interoperability becomes less of an issue between the client device and the cloud infrastructure – vendors are more interested in interoperability between their infrastructure components or with external service providers – via gateways.
This turned communications from a service into just another application on our phones.A new way to look at communication standards: WebRTC WebRTC brought communications to the browser, making it into a feature
WebRTC came to our world about 10 years ago and changed the paradigm again.
Where the smartphone and the cloud reduced our dependency and need for interoperability, WebRTC reduced our dependency and need for standardization.
We still need standardization – after all, WebRTC is a standard.
But the standardization we care about is mostly the browser implementations versus the specification (and interoperability between browsers). Other than that? We couldn’t care less.
If we need standardization and interoperability, we let gateways do it. As we did in the era of the smartphone.
WebRTC also made communications more accessible. Web developers could now use it, and you could easily embed and stitch it right into your application as a seamless part of your business process flow.
This turned communications from a service or an application into a feature in another service or application.Zoom and the pandemic The pandemic made video communication commonplace, enabling Zoom to turn it into a platform
Then the pandemic came and made a world a lot smaller. It made sure we all know how to use video communications.
Zoom became a household name across the globe and turned into a noun.
Zoom is proprietary. It doesn’t even use WebRTC.
No standards. Which lead to a lot of security missteps.
But it worked. And now Zoom has a Client SDK a Video SDK and Zoom Apps. With the intent of making their infrastructure and technology integratable with anything and everything.
This is an attempt to turn communications from a service or an application or a feature into… a platform.Workshop: The Future of Video & Realtime Communications
A few weeks ago, I had a conversation with Dean Bubley. We wanted to do something together, and decided to create a joint workshop.
The question of the role of standardization and interoperability is one of those we are going to tackle in the upcoming workshop.
If you are interested in joining the workshop, register below. There’s an early bird discount that is available only until the end of this month.REGISTER TO THE WORKSHOP
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There are new audio codecs in town: Google Lyra and Microsoft Satin. Both banking on AI-based voice coding, and both will be fighting for inclusion in WebRTC.
Right on the heels of the changes we see in video codecs in WebRTC, with AV1 coming into the stage, and HEVC making an entrance in Apple devices, we now have a similar (?) story with voice codecs. Microsoft announced its AI-powered voice codec Satin in February. A week later, Google reciprocated in kind, announcing its low bitrate codec for speech compression Lyra.
Why now? What are the similarities and differences between these codecs? Where are they headed? And what does that mean to WebRTC and to you?Table of contents
It makes sense to start this by explaining a bit about audio codecs in WebRTC.
For all intent and purposes G.711 is there as a legacy codec, to deal with narrowband audio. The result of which is low quality, unresilient audio. Using G.711 is mostly reserved to connect it with the telephony networks, and even there, I wouldn’t recommend it as a solution.
Opus is the main voice codec in WebRTC. It offers a highly flexible solution capable of handling anything from narrowband to fullband stereo and at low bitrates. You can read more in this article I’ve written years ago: The Rise of Opus to HD Voice Domination.
Opus is almost 10 years old. It has been created by meshing two separate codecs: SILK (for speech) and CELT (for music). The pandemic of 2020, and the increased reliance on virtual meetings has started to show its age and its limitations. Opus is a great codec, but these days, we can probably do better.How Opus works, from my Advanced WebRTC Architecture Course The two extremes in audio codecs: Low bitrate vs lossless
So what are the missing pieces? The things that Opus can’t get done on its own? There are two such areas that are actively being explored, and they are two extremes: highest possible audio quality and lowest possible bitrate.Highest possible quality: Lossless audio coding
One extreme (unrelated to Lyra and Satin), is the strive for the highest possible audio quality. Getting there requires the use of lossless audio coding.
For all intent and purpose, what we do in VoIP today, and by extension in WebRTC, is use lossy coding. This means that we compress the audio and video in ways that don’t really allow us to reconstruct the original audio or video accurately, but instead it gets us “close enough” to there. It does that by trying to “get rid” only of information that we humans can’t discern – things the human eye and human ear would miss anyways.
As a crude example, I never did hear the difference between vinyl, cassettes and CDs – at least not enough for it to matter for me. On the other hand, I had a friend who complained that CDs don’t have the audio quality of vinyl records.
The most known lossless audio codec is FLAC. It has nothing to do with WebRTC. Yet.Lowest possible bitrate: AI based compression
In the other end of this spectrum lies the lowest possible bitrate we can comfortably reach.
It turns out that Opus is good, but not great.
At a time where bandwidths are increasing, why do we even discuss getting voice codecs into lower and lower bitrates? What would be the incentive?
These questions are doubly important considering the fact that we’re heading towards a remote video filled world. And we know that video takes up considerably more bitrate than voice, so why care so much about voice bitrates?
One reason is simply the fact that we’re now communicating remotely a lot more. We do that more, and there are also a lot more people communicating online. From everywhere. This means that not all of them are going to be on great networks at all times, and even when they are, others are going to strain these networks with their own traffic. Google calls this “the next billion” – the next billion people joining the internet, which means people with less means and by extension less bandwidth.
The other reason is the fact that we’re growing bigger. More sessions. Bigger sessions. Widely spread. If we can even reduce a fraction of the bitrate, that would reduce the strain on our networks, servers and costs of running services.
I am also guessing that the big video meeting vendors got to learn a few interesting things during the pandemic. One of them is that voice is the most important part of a video call. If you don’t deliver your voice properly, video won’t matter. And for that, you need to make it leaner and meaner than it is today.
How do you make voice compression for an audio codec better?AI and audio codec generations
I’ll be using machine learning (ML) and artificial intelligence (AI) interchangeably here. These terms have been butchered by marketers so much, that they are now indistinguishable anyway.
Better in the case of audio codecs is going to be a new generation of codecs. In a way, a migration from the old way of doing things (rule engines and heuristics) to our brave new world of machine learning and artificial intelligence.
Machine learning is where the future lies when it comes to most of our algorithms. Especially with the ones that make extensive use today of either rule engines or heuristics – both of which are found in abundance in real time media processing pipelines (=WebRTC). We started seeing this trend seeping into real time communications and WebRTC somewhere in 2018. After the initial hype, we found out the many challenges of adding machine learning. In 2020, it seemed like the path became somewhat clearer: noise suppression and background replacement solutions assisted with AI. For the rest? We understood collectively that we should first squeeze the lemon of optimization before resorting to AI.
It is now time to look at AI in media compression as well. We’ve seen this take place already in baby steps. At Kranky Geek 2019, Shawn Zhong of Agora, explained how AI can be used to improve encoding efficiency:
A year later, NVIDIA introduced Maxine, a platform capable of using AI to “reconstruct” a person. Effectively creating a kind of a compression algorithm.
Research around AI compression is flourishing. There is already an AI specific standards organization called MPAI (Moving Picture, Audio and Data Coding by Artificial Intelligence) – still small, but this may change in the future. And then there’s Mozilla’s Common Voice, an open source, high quality, labeled multi-language dataset for training language related models.
It makes sense then, that audio would be a prime target for AI based compression as well. Here, Microsoft took the first public shot, and Google immediately followed suit.The Opus spec
To understand where Microsoft Satin and Google Lyra are headed, let’s first review how Opus works:
Now let’s look at what we know so far about the two new audio codecs.Microsoft Satin
Microsoft Satin is being positioned as an AI-powered audio codec to replace Silk.
Silk is used by Skype and was adopted as the basis for Opus as well. Here’s what Satin can do based on Microsoft’s announcement:
Satin wasn’t presented as a work in progress, but rather as a battle tested codec – Microsoft stated it is already being used by Microsoft Teams and Skype in 2-way calls. Obviously, with plans to extend it to group calls.
Satin is a brand new codec that is being designed to replace Opus altogether.Google Lyra
Google’s announcement of Lyra came a week after Microsoft’s. In a way, it seemed a bit rushed.
Why rushed? Because of how the announcement is written. It reads similar enough to the Microsoft one but lacks the “currently deployed” paragraph. Instead it has a “currently rolling out” paragraph.
What is Lyra about? Based on Google’s announcement:
Lyra is intended for SPEECH and not for AUDIO. It isn’t a replacement of Opus in any way.
Interestingly, Google believes that coupled with AV1, it can offer decent video conferencing experience at dial-in modem bitrates of 56kbps.
Lyra is being rolled out to Google Duo for very low bandwidth connections scenarios. But that’s about it for the time being.
More recently, Lyra has been open sourced by Google. The reasons for this are varied, especially considering that many of the recent advancements of Google in AI around real time communications weren’t open sourced at all:
Another thing you can achieve with Lyra is better redundancy for improved resiliency. With its very low bitrate, it is less of a constraint to add redundancy on top of it. You can check out this article on webrtcHacks by Philipp about audio redundancy encoding.A multi-codec audio future for WebRTC?
At the moment, both Lyra and Satin are nice bedtime stories. You can use them only inside the proprietary implementations of Google and Microsoft. And even then, in most cases you wouldn’t even know that to be the case.
Why was it important then to announce these efforts?
My hunch is that it has to do with standardization and WebRTC.
WebRTC needs some love and attention now in the audio front. For video, we’re going to have AV1, but what do we do about voice?
There are currently two alternatives out there that will make their move soon enough:
It is too early to say how this will play out. My bet is on more optional audio codecs finding their way into WebRTC – not the boring old ones, but rather the hip new ones. This will make audio codec selection for developers building services a wee bit harder, which isn’t a good thing in the long run. I’d rather see this pushed into Opus – or added as a single codec replacement to Opus. Something that would be easy to pick instead of Opus.FAQ on Satin and Lyra ✅ Is Google Lyra equivalent to Microsoft Satin?
While both of these audio codecs operate at low bitrates and are powered by AI they are very different. Lyra is focused on narrowband only while Satin is about operating in super wideband.
Technically – yes.
Microsoft is already using Satin instead of Opus in Microsoft Teams and Skype for 1:1 calls. IT was designed with that goal in mind.
Lyra was designed to work at low bitrates where Opus doesn’t do a good job today. When there’s enough bitrate, Opus offers better audio quality than Lyra.
There are no public plans to add either of these codecs to the WebRTC specification or to browser implementations.
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