Tag Archives: Latency

Cloudflare Stream Low-Latency HLS support now in Open Beta

Post Syndicated from Taylor Smith original http://blog.cloudflare.com/cloudflare-stream-low-latency-hls-open-beta/

Cloudflare Stream Low-Latency HLS support now in Open Beta

Cloudflare Stream Low-Latency HLS support now in Open Beta

Stream Live lets users easily scale their live-streaming apps and websites to millions of creators and concurrent viewers while focusing on the content rather than the infrastructure — Stream manages codecs, protocols, and bit rate automatically.

For Speed Week this year, we introduced a closed beta of Low-Latency HTTP Live Streaming (LL-HLS), which builds upon the high-quality, feature-rich HTTP Live Streaming (HLS) protocol. Lower latency brings creators even closer to their viewers, empowering customers to build more interactive features like chat and enabling the use of live-streaming in more time-sensitive applications like live e-learning, sports, gaming, and events.

Today, in celebration of Birthday Week, we’re opening this beta to all customers with even lower latency. With LL-HLS, you can deliver video to your audience faster, reducing the latency a viewer may experience on their player to as little as three seconds. Low Latency streaming is priced the same way, too: $1 per 1,000 minutes delivered, with zero extra charges for encoding or bandwidth.

Broadcast with latency as low as three seconds.

LL-HLS is an extension of the HLS standard that allows us to reduce glass-to-glass latency — the time between something happening on the broadcast end and a user seeing it on their screen. That includes factors like network conditions and transcoding for HLS and adaptive bitrates. We also include client-side buffering in our understanding of latency because we know the experience is driven by what a user sees, not when a byte is delivered into a buffer. Depending on encoder and player settings, broadcasters' content can be playing on viewers' screens in less than three seconds.

On the left, OBS Studio broadcasting from my personal computer to Cloudflare Stream. On the right, watching this livestream using our own built-in player playing LL-HLS with three second latency!

Same pricing, lower latency. Encoding is always free.

Our addition of LL-HLS support builds on all the best parts of Stream including simple, predictable pricing. You never have to pay for ingress (broadcasting to us), compute (encoding), or egress. This allows you to stream with peace of mind, knowing there are no surprise fees and no need to trade quality for cost. Regardless of bitrate or resolution, Stream costs \$1 per 1,000 minutes of video delivered and \$5 per 1,000 minutes of video stored, billed monthly.

Stream also provides both a built-in web player or HLS/DASH manifests to use in a compatible player of your choosing. This enables you or your users to go live using the same protocols and tools that broadcasters big and small use to go live to YouTube or Twitch, but gives you full control over access and presentation of live streams. We also provide access control with signed URLs and hotlinking prevention measures to protect your content.

Powered by the strength of the network

And of course, Stream is powered by Cloudflare's global network for fast delivery worldwide, with points of presence within 50ms of 95% of the Internet connected population, a key factor in our quest to slash latency. We ingest live video close to broadcasters and move it rapidly through Cloudflare’s network. We run encoders on-demand and generate player manifests as close to viewers as possible.

Getting started with LL-HLS

Getting started with Stream Live only takes a few minutes, and by using Live Outputs for restreaming, you can even test it without changing your existing infrastructure. First, create or update a Live Input in the Cloudflare dashboard. While in beta, Live Inputs will have an option to enable LL-HLS called “Low-Latency HLS Support.” Activate this toggle to enable the new pipeline.

Cloudflare Stream Low-Latency HLS support now in Open Beta

Stream will automatically provide the RTMPS and SRT endpoints to broadcast your feed to us, just as before. For the best results, we recommend the following broadcast settings:

  • Codec: h264
  • GOP size / keyframe interval: 1 second

Optionally, configure a Live Output to point to your existing video ingest endpoint via RTMPS or SRT to test Stream while rebroadcasting to an existing workflow or infrastructure.

Stream will automatically provide RTMPS and SRT endpoints to broadcast your feed to us as well as an HTML embed for our built-in player.

Cloudflare Stream Low-Latency HLS support now in Open Beta

This connection information can be added easily to a broadcast application like OBS to start streaming immediately:

Cloudflare Stream Low-Latency HLS support now in Open Beta

During the beta, our built-in player will automatically attempt to use low-latency for any enabled Live Input, falling back to regular HLS otherwise. If LL-HLS is being used, you’ll see “Low Latency” noted in the player.

During this phase of the beta, we are most closely focused on using OBS to broadcast and Stream’s built-in player to watch — which uses HLS.js under the hood for LL-HLS support. However, you may test the LL-HLS manifest in a player of your own by appending ?protocol=llhls to the end of the HLS manifest URL. This flag may change in the future and is not yet ready for production usage; watch for changes in DevDocs.

Sign up today

Low-Latency HLS is Stream Live’s latest tool to bring your creators and audiences together. All new and existing Stream subscriptions are eligible for the LL-HLS open beta today, with no pricing changes or contract requirements — all part of building the fastest, simplest serverless live-streaming platform. Join our beta to start test-driving Low-Latency HLS!

Workers KV is faster than ever with a new architecture

Post Syndicated from Charles Burnett original http://blog.cloudflare.com/faster-workers-kv-architecture/

Workers KV is faster than ever with a new architecture

Workers KV is faster than ever with a new architecture

We’re excited to announce a significant performance improvement coming to Workers KV, focused on dramatically improving cold read performance and reducing latency, even for long tail access patterns.

Developers using KV have seen great performance on hot reads, but ask why their 95th percentile latency — often on a key (or set of keys) that hadn’t been accessed recently or in that region — was higher than expected. We took this feedback to heart: we’ve been working feverishly on a new caching layer for KV behind the scenes, which enables customers to achieve much more frequent hot reads, reduced worst case latency times, better flexibility and control over cache TTLs, and much faster consistency over our previous iterations, and it’s now live for all KV users.

The best part? Developers using KV don’t need to change anything to benefit from this increased performance.

What is Workers KV?

Workers KV is a key value store designed for read heavy use-cases and applications powered by Cloudflare’s network. KV’s focus on read-heavy use-cases allows it to serve hot (cached) reads in milliseconds, which makes it ideal for storing per-application or customer configuration data, routing configuration, multivariate (A/B testing) configurations, and even small asset data that you need to serve quickly.  Anything that you can serialize and need quickly you can store in KV, all the way up to 25 MiB worth of data per each individual key, with no cap on total data stored.

The problem

KV might be optimized for read-heavy workloads, but it’s critical that writes are globally available quickly enough that they’re useful for your application. Under typical conditions, the convergence delay for an eventually consistent system like KV is approximately one minute, globally: a write from one location should be able to be observed by all readers. Typical conditions are great, but typical unfortunately didn’t mean “always”. It could take significant time to restore global consistency where regions like North America and Europe are reading the same value. We needed to improve not just the average convergence, but the worst case as well.

Speaking of consistency, setting a long cache Time to Live (cacheTTL) for reads would result in a situation where you won’t notice a write for the entire cacheTTL duration, as the existing cached data had not timed out yet. This means you have to trade off read latency for infrequently accessed keys against noticing writes. Developers using KV have been consistent in their feedback: a higher cache TTL should improve performance, but not multiply the time it takes for KV to converge on a write to that key.

Lastly, our cold read times also left room for improvement. While cache hits are fast in KV, a cache miss would result in a request being routed all the way to our storage backends. While this is slow for everyone, it was particularly slow for folks in regions not immediately in the US or EU.This is poor performance that doesn’t represent what we can achieve with our global presence.

Our solution

A new horizontally scaled tiered cache

We’ve revamped Workers KV to be powered by a new tiered cache implementation. This implementation is written as a Worker service. We reuse the Dynamic Dispatch infrastructure developed for Workers for Platforms which lets us jump from our old KV worker into our new caching service within hundreds of microseconds. Importantly, this means we don’t impact cache hit performance to implement this new transparent caching layer. We leverage the same infrastructure powering Smart Placement to implement the tiering.

Before we re-designed KV, our topology looked like this:

Workers KV is faster than ever with a new architecture
All data centers in Cloudflare’s network can reach out to the origin in the event of a cache miss or to do a background refresh.

Cache TTL and efficiency

Our design goal was clear and ambitious: “can we relax honoring the cacheTTL constraint without violating it”? While this seems contradictory, the motivation is clear: we want to minimize the need to communicate with our storage backends while honoring the user-facing semantics of the cacheTTL setting, as it can have security implications if violated (e.g. if you use it to store and validate security tokens). Answering this design question also manages to simultaneously solve many of the problems outlined earlier.

Comparing existing solutions

First, let’s look at the design constraints for two eventually consistent storage systems at Cloudflare: Quicksilver and Tiered CDN.

Quicksilver gives us global consistency within seconds using a push architecture to replicate the data across all machines at Cloudflare. That architecture however doesn’t scale for Workers KV’s needs, which can have terabytes of data just within one namespace. This would be too much to replicate to every single data center.

By comparison, the tiered CDN cache is a pull mechanism where each hop pulls a more recent version of the asset into the local cache on access. That scales better because we only use storage for assets that are accessed, which works well with most use-cases where the vast majority of data is never retrieved. However, a pull based architecture is insufficient because it can only let us aggregate traffic across broader regions but we still can’t decouple how long we serve from the cache from the cacheTTL.

Push based architectures let us know when an asset is updated and enable scalable storage. By blending the properties of both systems, we can decouple how long we store the assets in cache from the cacheTTL. And that’s exactly what we did: KV now uses a hybrid push/pull caching layer where data centers closest to customers will pull from the regional data centers that are a little bit farther away. Writes will broadcast to all regional data centers that a key has been updated, so that the regional data center will remove that key from the local cache.

Workers KV is faster than ever with a new architecture
Traditional regional tiered cache topology

We can solve this problem by taking advantage of the fact that we semantically understand the write operations that are happening within Workers KV:

  1. Workers KV doesn’t have one data center per region as might be typical for your zone in a Cloudflare CDN regional tiered cache topology. Instead, each key in a KV namespace is deterministically assigned a data center by performing a weighted rendezvous hash. The rendezvous hash ensures that load is distributed equally across the region and outages result in optimal shifts of traffic.
  2. When the data center closest to a customer has a miss, it computes the regional data center affinity and provides that information to our Smart Placement infrastructure. When a regional tier misses, we repeat this process except using data centers in the KV origin region.
  3. Finally, a miss at the upper tier exits to our storage nodes located in that origin region.

When we do a write, we only purge (invalidate) the key from the regional and upper tier data centers. This is a fixed number of data centers in our network regardless of how many data centers we add, which ensures that we aren’t reducing cache hit rates as our network continues to grow Compared with a global purge that delivers the event to every data center in our network, because we only need to deliver this purge to a random fixed set of data centers in our network, our aggregate write capacity for Workers KV automatically scales horizontally as we add more data centers.

Workers KV is faster than ever with a new architecture
All lower-tier data centers will reach out to a regional tier responsible for a given key in the event of a cache miss. If the regional tier doesn’t have the content, the regional tier will then ask an upper-tier out of region for the content. On a write for a given key, the responsible regional and upper tiers have that key deleted from cache.

Why do we call this a hybrid topology? The data centers closest to customers pull from the regional data centers as normal, but we automatically push invalidation events to the regional tier data centers on every write. That way, those customer data center pulls know to get an updated value when there is one. This means that while the cacheTTL parameter controls the caching behavior closest to the customer, it’s treated as a suggestion at best at the regional and upper tiers.

This way we’ve combined the push design principles behind Quicksilver, which delivers global consistency within seconds, with the pull-based design of our CDN tiered caching which can scale to handle “infinite” size workloads and prioritizes the assets that are most frequently accessed.

Visualizing it

It can be a bit hard to follow what’s happening in the new caching layer since there’s so many moving parts.

Here’s a video of a simplified version of how it works:

Small yellow balls represent KV read requests, larger green balls represent read responses. A larger purple ball represents a KV write request, while a read response ball represents a KV write response. Teal balls represent purge requests being broadcast. The “E” is a data center that doesn’t participate as a regional tier. The R represents the regional tier for key N while O is the upper tier for key N.

Decoupled cache TTL and consistency parameters

As a refresher, the objects written to KV can specify a cacheTTL: by default this is set to 1 minute, which is also the minimum acceptable value. This means that if an asset has been in the cache for longer than a minute, we bypass the cache and read instead from our durable storage nodes. In order to prevent eyeballs noticing origin fetches every minute, we implement stale while revalidate logic in our caching layer that automatically refreshes from the storage nodes in the background as requests come in.

Workers KV is faster than ever with a new architecture
Here’s an example from a Worker that’s constantly reading the same key

Notice the absence of any spikes indicating a cache miss? You’d expect to see them regularly every minute or so in the tens or even hundreds of milliseconds when the cacheTTL should expire. The reason this doesn’t happen is because as the expiry time is approaching, a background request to the storage nodes occurs and the cache is updated with an expiry time one more minute into the future; thus the asset in cache is never too stale and eyeball requests are always served from cache. Let’s take a look at requests to our storage layer before and after adding tiering:

Workers KV is faster than ever with a new architecture
Yellow is the estimated number of requests that would have occurred to origin without the new caching layer. Blue is the number of requests we’re making now.

The above chart is for a system with conservative parameters set. The upper tier doesn’t store the data for much longer than the cacheTTL currently and the upper tier will itself still do a background refresh probabilistically even though it doesn’t actually need to since we see all writes.

The new caching layer we’ve built inherits the old background refresh mechanism and expands on it. The first thing we did is decouple the background refresh period from the cacheTTL as a separate parameter (also defaulting to 1 minute). This means that even if you set a cacheTTL for 1 hour, KV will still check every minute from the regional tier to see if the value has been updated. If the data you’re storing within KV doesn’t have strict requirements on stale reads (think a key that’s accessed once every 10 minutes but needs to honor a write within 1 minute like security tokens), then you can increase the cacheTTL so that infrequently accessed keys stick around in the cache without changing the observed consistency.

Consistency improvements

Speaking of consistency, we’ve improved the worst case performance of that as well. Historically, we’ve had a background system that crawls all data in the storage nodes to figure out which region has the most up to date value and update accordingly. This gives us complete consistency coverage, but could take a significant amount of time to confirm. We would also periodically check both backends to see if network conditions had changed to pick the primary storage region to use for a given customer-close data center. Of course inconsistencies would be resolved then, but in practice this happens randomly, and at a low probability that won’t typically catch any meaningful values served inconsistently.

With the new caching layer all this changes. Since we’re now only reading keys on first access or after a write, we have enough storage capacity that we can check both backends on every read. When a customer requests data, we make a call to each origin data center, with the fastest response being returned immediately to reduce read latency. If the other data center has a newer value than what was returned first, we synchronize both data centers and notify our caching layer to purge that key from all regional data centers. If the other data center instead has an older value, we just synchronize the data centers without purging since we served the latest value. This means that even if our data centers are inconsistent, readers will notice new values much more quickly.

Latency improvements

Here’s the latency improvement at 10% rollout on a logarithmic x-axis:

Workers KV is faster than ever with a new architecture

Architecture that just gets better

This is just the start of what we can do. We now have a solid foundation for making further improvements, including making our best case reads even faster. We’ll be working on cutting out parts of our traditional stack that add unnecessary latency, and adding new high performance features that were too difficult to integrate otherwise. We can also explore features like setting the consistency TTL parameter for sub one minute consistency for additional cost. Similarly, we could create a best effort global purge feature if you want to choose to signal writes that way. Finally, we’re looking at exposing this new caching layer as a general Worker binding anyone can use within a Worker in front of their own service or to put in front of their Worker. If these sound like the killer features you need, please reach out to us if you’re interested in trying them out.

What next?

Developers don’t have to do anything to benefit from KV’s new performance improvements. We are currently in the process of rolling out our new architecture, and you don’t have to redeploy your Worker or change the way you use KV to benefit.

Workers KV is a natural fit for any application built on top of our Workers platform. We provide a native API that enables any Worker script to read, write, list, and manageyour Workers KV storage. You can also interact with Workers KV directly via our REST API from any client that can make a HTTP request, and the Cloudflare Dashboard provides an easy way to create, list, and delete keys to be used with the rest of your Workers setup.

Regardless of how you use Workers KV, it will be faster than ever before. We’re excited to see what you build with us, and you can dive into our documentation to start building with it.

Reduce latency and increase cache hits with Regional Tiered Cache

Post Syndicated from Alex Krivit original http://blog.cloudflare.com/introducing-regional-tiered-cache/

Reduce latency and increase cache hits with Regional Tiered Cache

Reduce latency and increase cache hits with Regional Tiered Cache

Today we’re excited to announce an update to our Tiered Cache offering: Regional Tiered Cache.

Tiered Cache allows customers to organize Cloudflare data centers into tiers so that only some “upper-tier” data centers can request content from an origin server, and then send content to “lower-tiers” closer to visitors. Tiered Cache helps content load faster for visitors, makes it cheaper to serve, and reduces origin resource consumption.

Regional Tiered Cache provides an additional layer of caching for Enterprise customers who have a global traffic footprint and want to serve content faster by avoiding network latency when there is a cache miss in a lower-tier, resulting in an upper-tier fetch in a data center located far away. In our trials, customers who have enabled Regional Tiered Cache have seen a 50-100ms improvement in tail cache hit response times from Cloudflare’s CDN.

What problem does Tiered Cache help solve?

First, a quick refresher on caching: a request for content is initiated from a visitor on their phone or computer. This request is generally routed to the closest Cloudflare data center. When the request arrives, we look to see if we have the content cached to respond to that request with. If it’s not in cache (it’s a miss), Cloudflare data centers must contact the origin server to get a new copy of the content.

Getting content from an origin server suffers from two issues: latency and increased origin egress and load.

Latency

Origin servers, where content is hosted, can be far away from visitors. This is especially true the more global of an audience a particular piece of content has relative to where the origin is located. This means that content hosted in New York can be served in dramatically different amounts of time for visitors in London, Tokyo, and Cape Town. The farther away from New York a visitor is, the longer they must wait before the content is returned. Serving content from cache helps provide a uniform experience to all of these visitors because the content is served from a data center that’s close.

Origin load

Even when using a CDN, many different visitors can be interacting with different data centers around the world and each data center, without the content visitors are requesting, will need to reach out to the origin for a copy. This can cost customers money because of egress fees origins charge for sending traffic to Cloudflare, and it places needless load on the origin by opening multiple connections for the same content, just headed to different data centers.

Reduce latency and increase cache hits with Regional Tiered Cache
When Tiered Cache is not enabled, all data centers in Cloudflare’s network can reach out to the origin in the event of a cache miss.

Performance improvements and origin load reductions are the promise of tiered cache.

Tiered Caching means that instead of every data center reaching out to the origin when there is a cache miss, the lower-tier data center that is closest to the visitor will reach out to a larger upper-tier data center to see if it has the requested content cached before the upper-tier asks the origin for the content. Organizing Cloudflare’s data centers into tiers means that fewer requests will make it back to the origin for the same content, preserving origin resources, reducing load, and saving the customer money in egress fees.

What options are there to maximize the benefits of tiered caching?

Cloudflare customers are given access to different Tiered Cache topologies based on their plan level. There are currently two predefined Tiered Cache topologies to select from – Smart and Generic Global. If either of those don’t work for a particular customer’s traffic profile, Enterprise customers can also work with us to define a custom topology.

In 2021, we announced that we’d allow all plans to access Smart Tiered Cache. Smart Tiered Cache dynamically finds the single closest data center to a customer’s origin server and chooses that as the upper-tier that all lower-tier data centers reach out to in the event of a cache miss. All other data centers go through that single upper-tier for content and that data center is the only one that can reach out to the origin. This helps to drastically boost cache hit ratios and reduces the connections to the origin. However, this topology can come at the cost of increased latency for visitors that are farther away from that single upper-tier.

Reduce latency and increase cache hits with Regional Tiered Cache
When Smart Tiered Cache is enabled, a single upper tier data center can communicate with the origin, helping to conserve origin resources.

Enterprise customers may select additional tiered cache topologies like the Generic Global topology which allows all of Cloudflare’s large data centers on our network (about 40 data centers) to serve as upper-tiers. While this topology may help reduce the long tail latencies for far-away visitors, it does so at the cost of increased connections and load on a customer's origin.

Reduce latency and increase cache hits with Regional Tiered Cache
When Generic Global Tiered Cache is enabled, lower-tier data centers are mapped to all upper-tier data centers in Cloudflare’s network which can all reach out to the origin in the event of a cache miss. 

To describe the latency problem with Smart Tiered Cache in more detail let’s use an example. Suppose the upper-tier data center is selected to be in New York using Smart Tiered Cache. The traffic profile for the website with the New York upper-tier is relatively global. Visitors are coming from London, Tokyo, and Cape Town. For every cache miss in a lower-tier it will need to reach out to the New York upper-tier for content. This means these requests from Tokyo will need to traverse the Pacific Ocean and most of the continental United States to check the New York upper-tier cache. Then turn around and go all the way back to Tokyo. This is a giant performance hit for visitors outside the US for the sake of improving origin resource load.

Regional Tiered Cache brings the best of both worlds

With Regional Tiered Cache we introduce a middle-tier in each region around the world. When a lower-tier fetches on a cache miss it tries the regional-tier first if the upper-tier is in a different region. If the regional-tier does not have the asset then it asks the upper-tier for it. On the response the regional-tier writes to its cache so other lower-tiers in the same region benefit.

By putting an additional tier in the same region as the lower-tier, there’s an increased chance that the content will be available in the region before heading to a far-away upper-tier. This can drastically improve the performance of assets while still reducing the number of connections that will eventually need to be made to the customer’s origin.

Reduce latency and increase cache hits with Regional Tiered Cache
When Regional Tiered Cache is enabled, all lower-tier data centers will reach out to a regional tier close to them in the event of a cache miss. If the regional tier doesn’t have the content, the regional tier will then ask an upper-tier out of region for the content. This can help improve latency for Smart and Custom Tiered Cache topologies.

Who will benefit from regional tiered cache?

Regional Tiered Cache helps customers with Smart Tiered Cache or a Custom Tiered Cache topology with upper-tiers in one or two regions. Regional Tiered Cache is not beneficial for customers with many upper-tiers in many regions like Generic Global Tiered Cache .

How to enable Regional Tiered Cache

Enterprise customers can enable Regional Tiered Cache via the Cloudflare Dashboard or the API:

UI

  • To enable Regional Tiered Cache, simply sign in to your account and select your website
  • Navigate to the Cache Tab of the dashboard, and select the Tiered Cache Section
  • If you have Smart or Custom Tiered Cache Topology Selected, you should have the ability to choose Regional Tiered Cache
Reduce latency and increase cache hits with Regional Tiered Cache

API

Please see the documentation for detailed information about how to configure Regional Tiered Cache from the API.

GET

curl --request GET \
 --url https://api.cloudflare.com/client/v4/zones/zone_identifier/cache/regional_tiered_cache \
 --header 'Content-Type: application/json' \
 --header 'X-Auth-Email: '

PATCH

curl --request PATCH \
 --url https://api.cloudflare.com/client/v4/zones/zone_identifier/cache/regional_tiered_cache \
 --header 'Content-Type: application/json' \
 --header 'X-Auth-Email: ' \
 --data '{
 "value": "on"
}'

Try Regional Tiered Cache out today!

Regional Tiered Cache is the first of many planned improvements to Cloudflare’s Tiered Cache offering which are currently in development. We look forward to hearing what you think about Regional Tiered Cache, and if you’re interested in helping us improve our CDN, we’re hiring.

Optimizing TCP for high WAN throughput while preserving low latency

Post Syndicated from Mike Freemon original https://blog.cloudflare.com/optimizing-tcp-for-high-throughput-and-low-latency/

Optimizing TCP for high WAN throughput while preserving low latency

Optimizing TCP for high WAN throughput while preserving low latency

Here at Cloudflare we’re constantly working on improving our service. Our engineers are looking at hundreds of parameters of our traffic, making sure that we get better all the time.

One of the core numbers we keep a close eye on is HTTP request latency, which is important for many of our products. We regard latency spikes as bugs to be fixed. One example is the 2017 story of “Why does one NGINX worker take all the load?”, where we optimized our TCP Accept queues to improve overall latency of TCP sockets waiting for accept().

Performance tuning is a holistic endeavor, and we monitor and continuously improve a range of other performance metrics as well, including throughput. Sometimes, tradeoffs have to be made. Such a case occurred in 2015, when a latency spike was discovered in our processing of HTTP requests. The solution at the time was to set tcp_rmem to 4 MiB, which minimizes the amount of time the kernel spends on TCP collapse processing. It was this collapse processing that was causing the latency spikes. Later in this post we discuss TCP collapse processing in more detail.

The tradeoff is that using a low value for tcp_rmem limits TCP throughput over high latency links. The following graph shows the maximum throughput as a function of network latency for a window size of 2 MiB. Note that the 2 MiB corresponds to a tcp_rmem value of 4 MiB due to the tcp_adv_win_scale setting in effect at the time.

Optimizing TCP for high WAN throughput while preserving low latency

For the Cloudflare products then in existence, this was not a major problem, as connections terminate and content is served from nearby servers due to our BGP anycast routing.

Since then, we have added new products, such as Magic WAN, WARP, Spectrum, Gateway, and others. These represent new types of use cases and traffic flows.

For example, imagine you’re a typical Magic WAN customer. You have connected all of your worldwide offices together using the Cloudflare global network. While Time to First Byte still matters, Magic WAN office-to-office traffic also needs good throughput. For example, a lot of traffic over these corporate connections will be file sharing using protocols such as SMB. These are elephant flows over long fat networks. Throughput is the metric every eyeball watches as they are downloading files.

We need to continue to provide world-class low latency while simultaneously providing high throughput over high-latency connections.

Before we begin, let’s introduce the players in our game.

TCP receive window is the maximum number of unacknowledged user payload bytes the sender should transmit (bytes-in-flight) at any point in time. The size of the receive window can and does go up and down during the course of a TCP session. It is a mechanism whereby the receiver can tell the sender to stop sending if the sent packets cannot be successfully received because the receive buffers are full. It is this receive window that often limits throughput over high-latency networks.

net.ipv4.tcp_adv_win_scale is a (non-intuitive) number used to account for the overhead needed by Linux to process packets. The receive window is specified in terms of user payload bytes. Linux needs additional memory beyond that to track other data associated with packets it is processing.

The value of the receive window changes during the lifetime of a TCP session, depending on a number of factors. The maximum value that the receive window can be is limited by the amount of free memory available in the receive buffer, according to this table:

tcp_adv_win_scale TCP window size
4 15/16 * available memory in receive buffer
3 ⅞ * available memory in receive buffer
2 ¾ * available memory in receive buffer
1 ½ * available memory in receive buffer
0 available memory in receive buffer
-1 ½ * available memory in receive buffer
-2 ¼ * available memory in receive buffer
-3 ⅛ * available memory in receive buffer

We can intuitively (and correctly) understand that the amount of available memory in the receive buffer is the difference between the used memory and the maximum limit. But what is the maximum size a receive buffer can be? The answer is sk_rcvbuf.

sk_rcvbuf is a per-socket field that specifies the maximum amount of memory that a receive buffer can allocate. This can be set programmatically with the socket option SO_RCVBUF. This can sometimes be useful to do, for localhost TCP sessions, for example, but in general the use of SO_RCVBUF is not recommended.

So how is sk_rcvbuf set? The most appropriate value for that depends on the latency of the TCP session and other factors. This makes it difficult for L7 applications to know how to set these values correctly, as they will be different for every TCP session. The solution to this problem is Linux autotuning.

Linux autotuning

Linux autotuning is logic in the Linux kernel that adjusts the buffer size limits and the receive window based on actual packet processing. It takes into consideration a number of things including TCP session RTT, L7 read rates, and the amount of available host memory.

Autotuning can sometimes seem mysterious, but it is actually fairly straightforward.

The central idea is that Linux can track the rate at which the local application is reading data off of the receive queue. It also knows the session RTT. Because Linux knows these things, it can automatically increase the buffers and receive window until it reaches the point at which the application layer or network bottleneck links are the constraint on throughput (and not host buffer settings). At the same time, autotuning prevents slow local readers from having excessively large receive queues. The way autotuning does that is by limiting the receive window and its corresponding receive buffer to an appropriate size for each socket.

The values set by autotuning can be seen via the Linux “ss” command from the iproute package (e.g. “ss -tmi”).  The relevant output fields from that command are:

Recv-Q is the number of user payload bytes not yet read by the local application.

rcv_ssthresh is the window clamp, a.k.a. the maximum receive window size. This value is not known to the sender. The sender receives only the current window size, via the TCP header field. A closely-related field in the kernel, tp->window_clamp, is the maximum window size allowable based on the amount of available memory. rcv_sshthresh is the receiver-side slow-start threshold value.

skmem_r is the actual amount of memory that is allocated, which includes not only user payload (Recv-Q) but also additional memory needed by Linux to process the packet (packet metadata). This is known within the kernel as sk_rmem_alloc.

Note that there are other buffers associated with a socket, so skmem_r does not represent the total memory that a socket might have allocated. Those other buffers are not involved in the issues presented in this post.

skmem_rb is the maximum amount of memory that could be allocated by the socket for the receive buffer. This is higher than rcv_ssthresh to account for memory needed for packet processing that is not packet data. Autotuning can increase this value (up to tcp_rmem max) based on how fast the L7 application is able to read data from the socket and the RTT of the session. This is known within the kernel as sk_rcvbuf.

rcv_space is the high water mark of the rate of the local application reading from the receive buffer during any RTT. This is used internally within the kernel to adjust sk_rcvbuf.

Earlier we mentioned a setting called tcp_rmem. net.ipv4.tcp_rmem consists of three values, but in this document we are always referring to the third value (except where noted). It is a global setting that specifies the maximum amount of memory that any TCP receive buffer can allocate, i.e. the maximum permissible value that autotuning can use for sk_rcvbuf. This is essentially just a failsafe for autotuning, and under normal circumstances should play only a minor role in TCP memory management.

It’s worth mentioning that receive buffer memory is not preallocated. Memory is allocated based on actual packets arriving and sitting in the receive queue. It’s also important to realize that filling up a receive queue is not one of the criteria that autotuning uses to increase sk_rcvbuf. Indeed, preventing this type of excessive buffering (bufferbloat) is one of the benefits of autotuning.

What’s the problem?

The problem is that we must have a large TCP receive window for high BDP sessions. This is directly at odds with the latency spike problem mentioned above.

Something has to give. The laws of physics (speed of light in glass, etc.) dictate that we must use large window sizes. There is no way to get around that. So we are forced to solve the latency spikes differently.

A brief recap of the latency spike problem

Sometimes a TCP session will fill up its receive buffers. When that happens, the Linux kernel will attempt to reduce the amount of memory the receive queue is using by performing what amounts to a “defragmentation” of memory. This is called collapsing the queue. Collapsing the queue takes time, which is what drives up HTTP request latency.

We do not want to spend time collapsing TCP queues.

Why do receive queues fill up to the point where they hit the maximum memory limit? The usual situation is when the local application starts out reading data from the receive queue at one rate (triggering autotuning to raise the max receive window), followed by the local application slowing down its reading from the receive queue. This is valid behavior, and we need to handle it correctly.

Selecting sysctl values

Before exploring solutions, let’s first decide what we need as the maximum TCP window size.

As we have seen above in the discussion about BDP, the window size is determined based upon the RTT and desired throughput of the connection.

Because Linux autotuning will adjust correctly for sessions with lower RTTs and bottleneck links with lower throughput, all we need to be concerned about are the maximums.

For latency, we have chosen 300 ms as the maximum expected latency, as that is the measured latency between our Zurich and Sydney facilities. It seems reasonable enough as a worst-case latency under normal circumstances.

For throughput, although we have very fast and modern hardware on the Cloudflare global network, we don’t expect a single TCP session to saturate the hardware. We have arbitrarily chosen 3500 mbps as the highest supported throughput for our highest latency TCP sessions.

The calculation for those numbers results in a BDP of 131MB, which we round to the more aesthetic value of 128 MiB.

Recall that allocation of TCP memory includes metadata overhead in addition to packet data. The ratio of actual amount of memory allocated to user payload size varies, depending on NIC driver settings, packet size, and other factors. For full-sized packets on some of our hardware, we have measured average allocations up to 3 times the packet data size. In order to reduce the frequency of TCP collapse on our servers, we set tcp_adv_win_scale to -2. From the table above, we know that the max window size will be ¼ of the max buffer space.

We end up with the following sysctl values:

net.ipv4.tcp_rmem = 8192 262144 536870912
net.ipv4.tcp_wmem = 4096 16384 536870912
net.ipv4.tcp_adv_win_scale = -2

A tcp_rmem of 512MiB and tcp_adv_win_scale of -2 results in a maximum window size that autotuning can set of 128 MiB, our desired value.

Disabling TCP collapse

Patient: Doctor, it hurts when we collapse the TCP receive queue.

Doctor: Then don’t do that!

Generally speaking, when a packet arrives at a buffer when the buffer is full, the packet gets dropped. In the case of these receive buffers, Linux tries to “save the packet” when the buffer is full by collapsing the receive queue. Frequently this is successful, but it is not guaranteed to be, and it takes time.

There are no problems created by immediately just dropping the packet instead of trying to save it. The receive queue is full anyway, so the local receiver application still has data to read. The sender’s congestion control will notice the drop and/or ZeroWindow and will respond appropriately. Everything will continue working as designed.

At present, there is no setting provided by Linux to disable the TCP collapse. We developed an in-house patch to the kernel to disable the TCP collapse logic.

Kernel patch – Attempt #1

The kernel patch for our first attempt was straightforward. At the top of tcp_try_rmem_schedule(), if the memory allocation fails, we simply return (after pred_flag = 0 and tcp_sack_reset()), thus completely skipping the tcp_collapse and related logic.

It didn’t work.

Although we eliminated the latency spikes while using large buffer limits, we did not observe the throughput we expected.

One of the realizations we made as we investigated the situation was that standard network benchmarking tools such as iperf3 and similar do not expose the problem we are trying to solve. iperf3 does not fill the receive queue. Linux autotuning does not open the TCP window large enough. Autotuning is working perfectly for our well-behaved benchmarking program.

We need application-layer software that is slightly less well-behaved, one that exercises the autotuning logic under test. So we wrote one.

A new benchmarking tool

Anomalies were seen during our “Attempt #1” that negatively impacted throughput. The anomalies were seen only under certain specific conditions, and we realized we needed a better benchmarking tool to detect and measure the performance impact of those anomalies.

This tool has turned into an invaluable resource during the development of this patch and raised confidence in our solution.

It consists of two Python programs. The reader opens a TCP session to the daemon, at which point the daemon starts sending user payload as fast as it can, and never stops sending.

The reader, on the other hand, starts and stops reading in a way to open up the TCP receive window wide open and then repeatedly causes the buffers to fill up completely. More specifically, the reader implemented this logic:

  1. reads as fast as it can, for five seconds
    • this is called fast mode
    • opens up the window
  2. calculates 5% of the high watermark of the bytes reader during any previous one second
  3. for each second of the next 15 seconds:
    • this is called slow mode
    • reads that 5% number of bytes, then stops reading
    • sleeps for the remainder of that particular second
    • most of the second consists of no reading at all
  4. steps 1-3 are repeated in a loop three times, so the entire run is 60 seconds

This has the effect of highlighting any issues in the handling of packets when the buffers repeatedly hit the limit.

Revisiting default Linux behavior

Taking a step back, let’s look at the default Linux behavior. The following is kernel v5.15.16.

Optimizing TCP for high WAN throughput while preserving low latency

The Linux kernel is effective at freeing up space in order to make room for incoming packets when the receive buffer memory limit is hit. As documented previously, the cost for saving these packets (i.e. not dropping them) is latency.

However, the latency spikes, in milliseconds, for tcp_try_rmem_schedule(), are:

tcp_rmem 170 MiB, tcp_adv_win_scale +2 (170p2):

@ms:
[0]       27093 |@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@
[1]           0 |
[2, 4)        0 |
[4, 8)        0 |
[8, 16)       0 |
[16, 32)      0 |
[32, 64)     16 |

tcp_rmem 146 MiB, tcp_adv_win_scale +3 (146p3):

@ms:
(..., 16)  25984 |@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@
[16, 20)       0 |
[20, 24)       0 |
[24, 28)       0 |
[28, 32)       0 |
[32, 36)       0 |
[36, 40)       0 |
[40, 44)       1 |
[44, 48)       6 |
[48, 52)       6 |
[52, 56)       3 |

tcp_rmem 137 MiB, tcp_adv_win_scale +4 (137p4):

@ms:
(..., 16)  37222 |@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@
[16, 20)       0 |
[20, 24)       0 |
[24, 28)       0 |
[28, 32)       0 |
[32, 36)       0 |
[36, 40)       1 |
[40, 44)       8 |
[44, 48)       2 |

These are the latency spikes we cannot have on the Cloudflare global network.

Kernel patch – Attempt #2

So the “something” that was not working in Attempt #1 was that the receive queue memory limit was hit early on as the flow was just ramping up (when the values for sk_rmem_alloc and sk_rcvbuf were small, ~800KB). This occurred at about the two second mark for 137p4 test (about 2.25 seconds for 170p2).

In hindsight, we should have noticed that tcp_prune_queue() actually raises sk_rcvbuf when it can. So we modified the patch in response to that, added a guard to allow the collapse to execute when sk_rmem_alloc is less than the threshold value.

net.ipv4.tcp_collapse_max_bytes = 6291456

The next section discusses how we arrived at this value for tcp_collapse_max_bytes.

The patch is available here.

The results with the new patch are as follows:

oscil – 300ms tests

Optimizing TCP for high WAN throughput while preserving low latency

oscil – 20ms tests

Optimizing TCP for high WAN throughput while preserving low latency

oscil – 0ms tests

Optimizing TCP for high WAN throughput while preserving low latency

iperf3 – 300 ms tests

Optimizing TCP for high WAN throughput while preserving low latency

iperf3 – 20 ms tests

Optimizing TCP for high WAN throughput while preserving low latency

iperf3 – 0ms tests

Optimizing TCP for high WAN throughput while preserving low latency

All tests are successful.

Setting tcp_collapse_max_bytes

In order to determine this setting, we need to understand what the biggest queue we can collapse without incurring unacceptable latency.

Optimizing TCP for high WAN throughput while preserving low latency
Optimizing TCP for high WAN throughput while preserving low latency

Using 6 MiB should result in a maximum latency of no more than 2 ms.

Cloudflare production network results

Current production settings (“Old”)

net.ipv4.tcp_rmem = 8192 2097152 16777216
net.ipv4.tcp_wmem = 4096 16384 33554432
net.ipv4.tcp_adv_win_scale = -2
net.ipv4.tcp_collapse_max_bytes = 0
net.ipv4.tcp_notsent_lowat = 4294967295

tcp_collapse_max_bytes of 0 means that the custom feature is disabled and that the vanilla kernel logic is used for TCP collapse processing.

New settings under test (“New”)

net.ipv4.tcp_rmem = 8192 262144 536870912
net.ipv4.tcp_wmem = 4096 16384 536870912
net.ipv4.tcp_adv_win_scale = -2
net.ipv4.tcp_collapse_max_bytes = 6291456
net.ipv4.tcp_notsent_lowat = 131072

The tcp_notsent_lowat setting is discussed in the last section of this post.

The middle value of tcp_rmem was changed as a result of separate work that found that Linux autotuning was setting receive buffers too high for localhost sessions. This updated setting reduces TCP memory usage for those sessions, but does not change anything about the type of TCP sessions that is the focus of this post.

For the following benchmarks, we used non-Cloudflare host machines in Iowa, US, and Melbourne, Australia performing data transfers to the Cloudflare data center in Marseille, France. In Marseille, we have some hosts configured with the existing production settings, and others with the system settings described in this post. Software used is perf3 version 3.9, kernel 5.15.32.

Throughput results

Optimizing TCP for high WAN throughput while preserving low latency

RTT (ms) Throughput with Current Settings (mbps) Throughput with New Settings (mbps) Increase Factor
Iowa to Marseille 121 276 6600 24x
Melbourne to Marseille 282 120 3800 32x

Iowa-Marseille throughput

Optimizing TCP for high WAN throughput while preserving low latency

Iowa-Marseille receive window and bytes-in-flight

Optimizing TCP for high WAN throughput while preserving low latency

Melbourne-Marseille throughput

Optimizing TCP for high WAN throughput while preserving low latency

Melbourne-Marseille receive window and bytes-in-flight

Optimizing TCP for high WAN throughput while preserving low latency

Even with the new settings in place, the Melbourne to Marseille performance is limited by the receive window on the Cloudflare host. This means that further adjustments to these settings yield even higher throughput.

Latency results

The Y-axis on these charts are the 99th percentile time for TCP collapse in seconds.

Cloudflare hosts in Marseille running the current production settings

Optimizing TCP for high WAN throughput while preserving low latency

Cloudflare hosts in Marseille running the new settings

Optimizing TCP for high WAN throughput while preserving low latency

The takeaway in looking at these graphs is that maximum TCP collapse time for the new settings is no worse than with the current production settings. This is the desired result.

Send Buffers

What we have shown so far is that the receiver side seems to be working well, but what about the sender side?

As part of this work, we are setting tcp_wmem max to 512 MiB. For oscillating reader flows, this can cause the send buffer to become quite large. This represents bufferbloat and wasted kernel memory, both things that nobody likes or wants.

Fortunately, there is already a solution: tcp_notsent_lowat. This setting limits the size of unsent bytes in the write queue. More details can be found at https://lwn.net/Articles/560082.

The results are significant:

Optimizing TCP for high WAN throughput while preserving low latency

The RTT for these tests was 466ms. Throughput is not negatively affected. Throughput is at full wire speed in all cases (1 Gbps). Memory usage is as reported by /proc/net/sockstat, TCP mem.

Our web servers already set tcp_notsent_lowat to 131072 for its sockets. All other senders are using 4 GiB, the default value. We are changing the sysctl so that 131072 is in effect for all senders running on the server.

Conclusion

The goal of this work is to open the throughput floodgates for high BDP connections while simultaneously ensuring very low HTTP request latency.

We have accomplished that goal.

Debugging High Latency Due to Context Leaks

Post Syndicated from Grab Tech original https://engineering.grab.com/debugging-high-latency-market-store

Background

Market-Store is an in-house developed general purpose feature store that is used to serve real-time computed machine learning (ML) features. Market-Store has a stringent SLA around latency, throughput, and availability as it empowers ML models, which are used in Dynamic Pricing and Consumer Experience.

Problem

As Grab continues to grow, introducing new ML models and handling increased traffic, Market-Store started to experience high latency. Market-Store’s SLA states that 99% of transactions should be within 200ms, but our latency increased to 2 seconds. This affected the availability and accuracy of our models that rely on Market-Store for real-time features.

Latency Issue

We used different metrics and logs to debug the latency issue but could not find any abnormalities that directly correlated to the API’s performance. We discovered that the problem went away temporarily when we restarted the service. But during the next peak period, the service began to struggle once again and the problem became more prominent as Market-Store’s query per second (QPS) increased.

The following graph shows an increase in the memory used with time over 12 hours. Even as the system load receded, memory usage continued to increase.

The continuous increase in memory consumption indicated the possibility of a memory leak, which occurs when memory is allocated but not returned after its use is over. This results in consistently increasing consumed memory until the service runs out of memory and crashes.

Although we could restart the service and resolve the issue temporarily, the increasing memory use suggested a deeper underlying root cause. This meant that we needed to conduct further investigation with tools that could provide deeper insights into the memory allocations.

Debugging Using Go Tools

PPROF is a profiling tool by Golang that helps to visualise and analyse profiles from Go programmes. A profile is a collection of stack traces showing the call sequences in your programme that eventually led to instances of a particular event i.e. allocation. It also provides details such as Heap and CPU information, which could provide insights into the bottlenecks of the Go programme.

By default, PPROF is enabled on all Grab Go services, making it the ideal tool to use in our scenario. To understand how memory is allocated, we used PPROF to generate Market-Store’s Heap profile, which can be used to understand how inuse memory was allocated for the programme.

You can collect the Heap profile by running this command:

go tool pprof 'http://localhost:6060/debug/pprof/heap'

The command then generates the Heap profile information as shown in the diagram below:

From this diagram, we noticed that a lot of memory was allocated and held by the child context created from Async Library even after the tasks were completed.

In Market-Store, we used the Async Library, a Grab open-source library, which typically used to run concurrent tasks. Any contexts created by the Async Library should be cleaned up after the background tasks are completed. This way, memory would be returned to the service.

However, as shown in the diagram, memory was not being returned, resulting in a memory leak, which explains the increasing memory usage even as Market-Store’s system load decreased.

Uncovering the Real Issue

So we knew that Market-Store’s latency was affected, but we didn’t know why. From the first graph, we saw that memory usage continued to grow even as Market-Store’s system load decreased. Then, PPROF showed us that the memory held by contexts was not cleaned up, resulting in a memory leak.

Through our investigations, we drew a correlation between the increase in memory usage and a degradation in the server’s API latency. In other words, the memory leak resulted in a high memory consumption and eventually, caused the latency issue.

However, there was no change in our service that would have impacted how contexts are created and cleaned up. So what caused the memory leak?

Debugging the Memory Leak

We needed to look into the Async Library and how it worked. For Market-Store, we updated the cache asynchronously for the write-around caching mechanism. We use the Async Library for running the update tasks in the background.

The following code snippet explains how the Async Library works:


async.Consume(context.Background(), runtime.NumCPU()*4, buffer)

// Consume runs the tasks with a specific max concurrency

func Consume(ctx context.Context, concurrency int, tasks chan Task) Task {

   // code...

   return Invoke(ctx, func(context.Context) (interface{}, error) {

       workers := make(chan int, concurrency)

       concurrentTasks := make([]Task, concurrency)

       // code ...

       t.Run(ctx).ContinueWith(ctx, func(interface{}, error) (interface{}, error) {

       // code...

      })

    }

}

func Invoke(ctx context.Context, action Work) Task {

    return NewTask(action).Run(ctx)

}

func(t *task) Run(ctx context.Context) Task {

    ctx, t.cancel = context.WithCancel(ctx)

    go t.run(ctx)

    return t

}

Note: Code that is not relevant to this article was replaced with code.

As seen in the code snippet above, the Async Library initialises the Consume method with a background context, which is then passed to all its runners. Background contexts are empty and do not track or have links to child contexts that are created from them.

In Market-Store, we use background contexts because they are not bound by request contexts and can continue running even after a request context is cleaned up. This means that once the task has finished running, the memory consumed by child contexts would be freed up, avoiding the issue of memory leaks altogether.

Identifying the Cause of the Memory Leak

Upon further digging, we discovered an MR that was merged into the library to address a task cancellation issue. As shown in the code snippet below, the Consume method had been modified such that task contexts were being passed to the runners, instead of the empty background contexts.

func Consume(ctx context.Context, concurrency int, tasks chan Task) Task {

     // code...

     return Invoke(ctx, func(taskCtx context.Context) (interface{}, error) {

         workers := make(chan int, concurrency)

         concurrentTasks := make([]Task, concurrency)

         // code ...

         t.Run(taskCtx).ContinueWith(ctx, func(interface{}, error) (interface{}, error) {

            // code...

        })

     }

}

Before we explain the code snippet, we should briefly explain what Golang contexts are. A context is a standard Golang package that carries deadlines, cancellation signals, and other request-scoped values across API boundaries and between processes. We should always remember to cancel contexts after using them.

Importance of Context Cancellation

When a context is cancelled, all contexts derived from it are also cancelled. This means that there will be no unaccounted contexts or links and it can be achieved by using the Async Library’s CancelFunc.

The Async Library’s CancelFunc method will:

  • Cancel the created child context and its children
  • Remove the parent reference from the child context
  • Stop any associated timers

We should always make sure to call the CancelFunc method after using contexts, to ensure that contexts and memory are not leaked.

Explaining the Impact of the MR

In the previous code snippet, we see that task contexts are passed to runners and they are not being cancelled. The Async Library created task contexts from non-empty contexts, which means the task contexts are tracked by the parent contexts. So, even if the work associated with these task contexts is complete, they will not be cleaned up by the system (garbage collected).

As we started using task contexts instead of background contexts and did not cancel them, the memory used by these contexts was never returned, thus resulting in a memory leak.

It took us several tries to debug and investigate the root cause of Market-Store’s high latency issue and through this incident, we learnt several important things that would help prevent a memory leak from recurring.

  • Always cancel the contexts you’ve created. Leaving it to garbage collection (system cleanup) may result in unexpected memory leaks.

  • Go profiling can provide plenty of insights about your programme, especially when you’re not sure where to start troubleshooting.

  • Always benchmark your dependencies when integrating or updating the versions to ensure they don’t have any performance bottlenecks.


Special thanks to Chip Dong Lim for his contributions and for designing the GIFs included in this article.


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