Slot Machine RTP: What Every Player Should Know |

Slot Machine RTP: What Every Player Should Know

This article was originally published on Feb 25, 2025 and was last updated on Feb 25, 2025.

If you’ve ever played slot games, you’ve likely come across the term “RTP.” It’s one of the most important elements of a slot game, and understanding it will help you make smarter decisions when choosing which title to play.

It offers insights into the potential returns a game can offer and helps you find your way around countless slot games offered by casinos. This guide will help you decode what RTP means, ultimately helping you make the most of your bets.

What is RTP?

RTP, or Return to Player, is a measure of the percentage of total wagers that a slot game is designed to pay back to players over time.

If a slot has an RTP of 96%, it means the game will return $96 for every $100 wagered, on average.

It’s important to remember that RTP is theoretical and applies to millions of spins, not individual gaming sessions.

Why Does It Matter So Much In Gambling?

RTP is a critical factor in gambling because it gives you an idea of how “player-friendly” a game is. Slots with higher RTPs generally offer better long-term returns, making them more attractive to players who value consistency.
While it doesn’t guarantee short-term success, understanding RTP can help you make informed choices and manage your bankroll more effectively. Remember that high RTP slots are ideal if you’re looking for better odds.

RTP vs. RNG

While RTP represents the theoretical return to players, RNG (Random Number Generator) ensures the fairness and unpredictability of each spin. The RNG is a software algorithm that generates random outcomes for every spin, ensuring no two outcomes are the same.
This randomness means the slot adheres to its stated RTP but does not guarantee when or how payouts occur. Together, RTP and RNG work to balance fairness with profitability, creating an exciting gaming experience.

How RTP Works – Key Aspects

To become wiser in your gambling journey, it’s also necessary to do your homework and do some math by yourself. Sometimes, knowing how to calculate the RTP of a slot game can help you figure out if that game is worth your betting money or not.

Formula to Calculate RTP

The RTP of a slot is calculated using a simple formula.

Here’s how it’s done:

(Total amount returned to players / Total amount wagered by players) x 100.

If a slot machine returns $9,600 from $10,000 in bets, its RTP is 96%. Because you divide 9,600 by 10,000, you get 0,96. Multiply that by 100 and you’ll get 96% RTP for that slot machine.

Game developers use advanced simulations over millions of spins to determine this value.

Impact on a Player’s Gaming Session

RTP directly influences the longevity of your gaming session. A higher RTP can mean longer playtime for the same bankroll, as you’re more likely to recover a larger percentage of your wagers. Conversely, slots with lower RTPs may lead to faster losses, especially over short gaming sessions. While RTP doesn’t guarantee individual outcomes, choosing a higher RTP game can give you a statistical edge over time.

House Edge vs RTP

The house edge is the flip side of RTP. If a slot has an RTP of 96%, the house edge is 4%, representing the casino’s statistical advantage. This percentage shows how much the casino expects to retain from player wagers over time. Understanding this relationship helps you see how RTP affects your odds as a player.

Types of Slots Based on RTP

Slots can be categorized based on their RTP percentages.

• High RTP Slots (97.5% and above) – They are the most favorable for players and are often considered low-risk options.
• Above Average RTP Slots (96.5% – 97.49%) – These slots have a slightly better payout rate than most games, making them a solid choice for players who prefer higher returns.
• Average RTP Slots (95% – 96.49%) – These are the industry standard and offer a balance between payouts and features.
• Below Average RTP Slots (94% – 94.99%) – These slots have a slightly lower payout rate, meaning the house edge is higher. You may experience more frequent losses compared to higher RTP slots.
• Low RTP Slots (Below 94%) – Slots with the lowest return percentages. These are considered high-risk games where the house edge is significant, and you should expect longer losing streaks.

Where to Find a Slot’s RTP

You can usually find a slot’s RTP in the game’s information or help section. Online casinos often display RTP in the game description or under technical specifications.

Many review sites and casino aggregators also list RTP for popular slots, allowing you to compare games before playing.

If you’re unsure, don’t hesitate to ask the casino’s support team or refer to official sources like game developer websites.

Even though the developer may release a title with a certain RTP, the casinos featuring that game may also be allowed to adjust the original RTP percentage.

The safest thing to do is check the information released by the casino pertaining to that specific game, and not base your decision solely on the RTP indicated by the game’s developers.

RTP and Volatility

While RTP determines the percentage of wagers returned to players over time, slot volatility refers to the frequency and size of payouts.

Low-volatility slots offer smaller, more frequent wins, making them ideal for casual players.

High-volatility slots, on the other hand, have less frequent payouts but they tend to be larger, appealing to risk-takers.

Understanding both factors is crucial, as they affect your overall experience and strategy. A game with a high RTP and low volatility might suit you if you prefer steady returns, while a high-volatility game with a lower RTP might appeal to those chasing big wins.

Land-Based vs Online Slots RTP

Land-based slots generally have lower RTPs, averaging between 85% and 90%. In contrast, online slots typically offer RTPs of 92% to 98%, giving players a better chance of long-term returns. The higher RTPs in online slots are largely due to lower operational costs, allowing developers to allocate more payouts to players.

How to Choose a Slot Game Based on RTP

When going for any slot game, you want to aim for the highest RTP percentages, so go above 96% whenever possible.

The easiest way to do so is to go to our dedicated page for the best payout slots. It displays a comprehensive list of titles with the highest possible RTP, displayed in descending order.

Next, consider the game’s volatility and align it with your risk tolerance. Make sure you read my dedicated guide on slot volatility so you can better understand volatility levels and how they work. (low-volatility games for smaller, more frequent payouts or high-volatility titles for the opposite effect).

Finally, examine the game’s paytable to see if it matches your gameplay preferences.

If you want to further customize the list and find the right slot game for you, use our advanced filter featured on the best payout slots page. This way, our huge catalog will display only the games corresponding to your preferences. Then, you’re free to further customize your selection, such as volatility, reels, software, and themes.

Top 10 Slots with High RTP

Not sure where to start with high RTP slot games? I’ve picked 10 titles featuring the highest Return to Player percentages out there. Feel free to experiment with any of these ones in demo mode before moving on to the real deal.

Common Myths About RTP

Throughout history, slot games have become enshrouded in myths and legends that can either make one hopeful (and also misinformed), or make one prone to simple mistakes. RTP is no exception.

With this in mind, here are the most prevalent myths and misconceptions about RTP levels in slot games.

One common myth is that RTP guarantees short-term results. In reality, RTP reflects long-term averages and doesn’t predict individual outcomes.

Another misconception is that high RTP slots always yield big wins. While they provide better long-term returns, factors like volatility and bonus features also play a role.

Lastly, many believe that all slots with the same RTP perform similarly. However, the gameplay experience varies widely based on other factors.

Final Thoughts

RTP is an essential concept for any slot player, providing insight into how much a game is likely to return over time.

From my perspective, understanding RTP is no longer just a technical aspect of gambling. Once you get the know-how of this core mechanic, you’ll soon be enjoying slot games in a smarter, more informed way. While it’s tempting to chase flashy games or massive jackpots, focusing on factors like RTP and volatility allows you to balance fun with strategy.

My advice to you? Go for slots with high RTP with medium volatility. This is the sweet spot, as they offer enough excitement while keeping your bankroll intact for longer.

Slots RTP FAQs

Does RTP vary between spins, or is it fixed?

RTP is a theoretical percentage calculated over millions of spins and remains constant for a specific game. However, your results in a single session may differ significantly because outcomes are determined by the RNG.

Are bonuses and free spins affected by a slot’s RTP?

Yes, the RTP applies to all gameplay, including free spins and bonus rounds. However, some bonus-specific features may have different probabilities or payout rates, so always check the game rules.

Can RTP be different for the same slot at different casinos?

Yes, some developers offer adjustable RTP ranges, allowing casinos to set their preferred percentage. Always verify the RTP on the platform where you’re playing, as it might differ from reviews or developer specifications

Was this article helpful?

Real-time Transport Protocol (RTP)

The Real-time Transport Protocol (RTP) is a network protocol that provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or unicast network services.

In this article:

• RTP Basics
• RTP Packet Header Fields
  • Payload Type
  • Sequence Number
  • Timestamp
  • Synchronization Source Identifier (SSRC)
  • RTCP Packet Types
    • Receiver reception packets
    • Sender report packets
    • Source description packets
    • Audio and Video Compression
    • H.323 Channels
    • Gatekeepers

    RTP Basics

    Real-time Transport Protocol (RTP) runs on top of UDP. Specifically, audio or video chunks of data, generated by the sending side of a multimedia application, are encapsulated in RTP packets, and each RTP packet is in turn encapsulated in a UDP segment.

    Because RTP provides services like timestamps or sequence numbers, to the multimedia application, RTP can be viewed as a sublayer of the transport layer.

    From the application developer’s perspective, however, RTP is not part of the transport layer but instead part of the application layer. This is because the developer must integrate RTP into the application. Specifically, for the sender side of the application, the developer must write code into the application which creates the RTP encapsulating packets; the application then sends the RTP packets into a UDP socket interface. Similarly, at the receiver side of the application, the RTP packets enter the application through a UDP socket interface; the developer therefore must write code into the application that extracts the media chunks from the RTP packets.

    

    If an application incorporates RTP — instead of a proprietary scheme to provide payload type, sequence numbers or timestamps – then, the application will more easily interoperate with other networking applications. For example, if two different companies develop Internet phone software, and they both incorporate RTP into their product, there may be some hope that a user using one of the Internet phone products will be able to communicate with a user using the other Internet phone product.

    It should be emphasized that RTP in itself does not provide any mechanism to ensure timely delivery of data or provide other quality of service guarantees; it does not even guarantee delivery of packets or prevent out-of-order delivery of packets. Indeed, RTP encapsulation is only seen at the end systems — it is not seen by intermediate routers. Routers do not distinguish between IP datagrams that carry RTP packets and IP datagrams that don’t.

    RTP allows each source (for example, a camera or a microphone) to be assigned its own independent RTP stream of packets. For example, for a videoconference between two participants, four RTP streams could be opened: two streams for transmitting the audio (one in each direction) and two streams for the video (again, one in each direction). However, many popular encoding techniques — including MPEG1 and MPEG2 — bundle the audio and video into a single stream during the encoding process. When the audio and video are bundled by the encoder, then only one RTP stream is generated in each direction.

    RTP packets are not limited to unicast applications. They can also be sent over one-to-many and many-to-many multicast trees. For a many-to-many multicast session, all of the senders and sources in the session typically send their RTP streams into the same multicast tree with the same multicast address. RTP multicast streams belonging together, such as audio and video streams emanating from multiple senders in a videoconference application, belong to an RTP session.

    RTP Packet Header Fields

    The four principal packet header fields are the payload type, sequence number, timestamp, and the source identifier.

    

    Payload Type

    The payload type field in the RTP packet is seven-bits long. Thus 2^7 or 128 different payload types can be supported by RTP. For an audio stream, the payload type field is used to indicate the type of audio encoding (e.g., PCM, adaptive delta modulation, linear predictive encoding) that is being used. If a sender decides to change the encoding in the middle of a session, the sender can inform the receiver of the change through this payload type field. The sender may want to change the encoding in order to increase the audio quality or to decrease the RTP stream bit rate.

    

    For a video stream the payload type can be used to indicate the type of video encoding (e.g., motion JPEG, MPEG1, MPEG2, H.231). Again, the sender can change video encoding on-the-fly during a session.

    

    Sequence Number Field

    The sequence number field is 16-bits long. The sequence number increments by one for each RTP packet sent, and may be used by the receiver to detect packet loss and to restore packet sequence. For example, if the receiver side of the application receives a stream of RTP packets with a gap between sequence numbers 86 and 89, then the receiver knows that packets 87 and 88 were lost. The receiver can then attempt to conceal the lost data.

    Timestamp Field

    The timestamp field is 32 bytes long. It reflects the sampling instant of the first byte in the RTP data packet. As we saw in the previous section, the receiver can use the timestamps in order to remove packet jitter introduced in the network and to provide synchronous playout at the receiver. The timestamp is derived from a sampling clock at the sender. As an example, for audio the timestamp clock increments by one for each sampling period (for example, each 125 usecs for a 8 KHz sampling clock); if the audio application generates chunks consisting of 160 encoded samples, then the timestamp increases by 160 for each RTP packet when the source is active. The timestamp clock continues to increase at a constant rate even if the source is inactive.

    Synchronization Source Identifier (SSRC)

    The SSRC field is 32 bits long. It identifies the source of the RTP stream. Typically, each stream in a RTP session has a distinct SSRC. The SSRC is not the IP address of the sender, but instead a number that the source assigns randomly when the new stream is started. The probability that two streams get assigned the same SSRC is very small.

    RTP Control Protocol (RTCP)

    Request For Comments 1889 also specifies RTCP, a protocol which a multimedia networking application can use in conjunction with RTP. The use of RTCP is particularly attractive when the networking application multicasts audio or video to multiple receivers from one or more senders.

    

    RTCP packets are transmitted by each participant in an RTP session to all other participants in the session. The RTCP packets are distributed to all the participants using IP multicast. For an RTP session, typically there is a single multicast address, and all RTP and RTCP packets belonging to the session use the multicast address. RTP and RTCP packets are distinguished from each other through the use of distinct port numbers.

    RTCP packets do not encapsulate chunks of audio or video. Instead, RTCP packets are sent periodically and contain sender and/or receiver reports that announce statistics that can be useful to the application. These statistics include number of packets sent, number of packets lost and interarrival jitter. The RTP specification [RFC 1889] does not dictate what the application should do with this feedback information. It is up to the application developer to decide what it wants to do with the feedback information. Senders can use the feedback information, for example, to modify their transmission rates. The feedback information can also be used for diagnostic purposes; for example, receivers can determine whether problems are local, regional or global.

    RTCP Packet Types

    Receiver reception packets

    For each RTP stream that a receiver receives as part of a session, the receiver generates a reception report. The receiver aggregates its reception reports into a single RTCP packet. The packet is then sent into multicast tree that connects together all the participants in the session. The reception report includes several fields, the most important of which are listed below.

    • The SSRC of the RTP stream for which the reception report is being generated.
    • The fraction of packets lost within the RTP stream. Each receiver calculates the number of RTP packets lost divided by the number of RTP packets sent as part of the stream. If a sender receives reception reports indicating that the receivers are receiving only a small fraction of the sender’s transmitted packets, the sender can switch to a lower encoding rate, thereby decreasing the congestion in the network, which may improve the reception rate.
    • The last sequence number received in the stream of RTP packets.
    • The interarrival jitter, which is calculated as the average interarrival time between successive packets in the RTP stream.

    Sender report packets

    For each RTP stream that a sender is transmitting, the sender creates and transmits RTCP sender-report packets. These packets include information about the RTP stream, including:

    • The SSRC of the RTP stream.
    • The timestamp and wall-clock time of the most recently generated RTP packet in the stream
    • The number of packets sent in the stream.
    • The number of bytes sent in the stream.

    The sender reports can be used to synchronize different media streams within a RTP session. For example, consider a videoconferencing application for which each sender generates two independent RTP streams, one for video and one for audio. The timestamps in these RTP packets are tied to the video and audio sampling clocks, and are not tied to the wall-clock time (i.e., to real time). Each RTCP sender-report contains, for the most recently generated packet in the associated RTP stream, the timestamp of the RTP packet and the wall-clock time for when the packet was created. Thus, the RTCP sender-report packets associate the sampling clock to the real-time clock. Receivers can use this association in the RTCP sender reports to synchronize the playout of audio and video.

    Source description packets

    For each RTP stream that a sender is transmitting, the sender also creates and transmits source-description packets. These packets contain information about the source, such as e-mail address of the sender, the sender’s name and the application that generates the RTP stream. It also includes the SSRC of the associated RTP stream. These packets provide a mapping between the source identifier (i.e., the SSRC) and the user/host name.

    RTCP packets are stackable, i.e., receiver reception reports, sender reports, and source descriptors can be concatenated into a single packet. The resulting packet is then encapsulated into a UDP segment and forwarded into the multicast tree.

    RTCP Bandwidth Scaling

    The astute reader will have observed that RTCP has a potential scaling problem. Consider for example an RTP session that consists of one sender and a large number of receivers. If each of the receivers periodically generate RTCP packets, then the aggregate transmission rate of RTCP packets can greatly exceed the rate of RTP packets sent by the sender. Observe that the amount of traffic sent into the multicast tree does not change as the number of receivers increases, whereas the amount of RTCP traffic grows linearly with the number of receivers. To solve this scaling problem, RTCP modifies the rate at which a participant sends RTCP packets into the multicast tree as a function of the number of participants in the session.

    Observe that, because each participant sends control packets to everyone else, each participant can keep track of the total number of participants in the session.

    RTCP attempts to limit its traffic to 5% of the session bandwidth. For example, suppose there is one sender, which is sending video at a rate of 2 Mbps. Then RTCP attempts to limit its traffic to 5% of 2 Mbps, or 100 Kbps, as follows. The protocol gives 75% of this rate, or 75 Kbps, to the receivers; it gives the remaining 25% of the rate, or 25 Kbps, to the sender. The 75 Kbps devoted to the receivers is equally shared among the receivers. Thus, if there are R receivers, then each receiver gets to send RTCP traffic at a rate of 75/R Kbps and the sender gets to send RTCP traffic at a rate of 25 Kbps. A participant (a sender or receiver) determines the RTCP packet transmission period by dynamically calculating the average RTCP packet size (across the entire session) and dividing the average RTCP packet size by its allocated rate.

    In summary, the period for transmitting RTCP packets for a sender is

    

    And the period for transmitting RTCP packets for a receiver is

    

    H.323

    H.323 is a standard for real-time audio and video conferencing among end systems on the Internet. As shown in Figure 6.4-7, it also covers how end systems attached to the Internet communicate with telephones attached to ordinary circuit-switched telephone networks. In principle, if manufacturers of Internet telephony and video conferencing all conform to H.323, then all their products should be able to interoperate and should be able to communicate with ordinary telephones. We discuss H.323 in this section, as it provides an application context for RTP. Indeed, we shall see below that RTP is an integral part of the H.323 standard.

    

    H.323 end points (a.k.a. terminals) can be stand-alone devices (e.g., Web phones and Web TVs) or applications in a PC (e.g., Internet phone or video conferencing software). H.323 equipment also includes gateways and gatekeepers. Gateways permit communication among H.323 end points and ordinary telephones in a circuit-switched telephone network. Gatekeepers, which are optional, provide address translation, authorization, bandwidth management, accounting and billing. We will discuss gatekeepers in more detail at the end of this section.

    The H.323 is an umbrella specification that includes:

    

    • A specification for how endpoints negotiate common audio/video encodings. Because H.323 supports a variety of audio and video encoding standards, a protocol is needed to allow the communicating endpoints to agree on a common encoding.
    • A specification for how audio and video chunks are encapsulated and sent over the network. As you may have guessed, this is where RTP comes into the picture.
    • A specification for how endpoints communicate with their respective gatekeepers.
    • A specification for how Internet phones communicate through a gateway with ordinary Phones in the public circuit-switched telephone network.

    Minimally, each H.323 endpoint must support the G.711 speech compression standard. G.711 uses PCM to generate digitized speech at either 56 kbps or 64 kbps. Although H.323 requires every endpoint to be voice capable (through G.711), video capabilities are optional. Because video support is optional, manufacturers of terminals can sell simpler speech terminals as well as more complex terminals that support both audio and video.

    H.323 also requires that all H.323 end points use the following protocols:

    • RTP – the sending side of an endpoint encapsulates all media chunks within RTP packets. Sending side then passes the RTP packets to UDP.
    • H.245 – an “out-of-band” control protocol for controlling media between H.323 endpoints. This protocol is used to negotiate a common audio or video compression standard that will be employed by all the participating endpoints in a session.
    • Q.931 – a signaling protocol for establishing and terminating calls. This protocol provides traditional telephone functionality (e.g., dial tones and ringing) to H.323 endpoints and equipment.
    • RAS (Registration/Admission/Status) channel protocol – a protocol that allows end points to communicate with a gatekeeper (if gatekeeper is present).

    Audio and Video Compression

    The H.323 standard supports a specific set of audio and video compression techniques. Let’s first consider audio. As we just mentioned, all H.323 end points must support the G.711 speech encoding standard. Because of this requirement, two H.323 end points will always be able to default to G.711 and communicate. But H.323 allows terminals to support a variety of other speech compression standards, including G.723.1, G.722, G.728 and G.729. Many of these standards compress speech to rates that will pass through 28.8 Kbps dial-up modems. For example, G.723.1 compresses speech to either 5.3 kbps or 6.3 kbps, with sound quality that is comparable to G.711.

    As we mentioned earlier, video capabilities for an H.323 endpoint are optional. However, if an endpoint does supports video, then it must (at the very least) support the QCIF H.261 (176×144 pixels) video standard. A video capable endpoint my optionally support other H.261 schemes, including CIF, 4CIF and 16CIF., and the H.263 standard. As the H.323 standard evolves, it will likely support a longer list of audio and video compression schemes.

    H.323 Channels

    When a end point participates in an H.323 session, it maintains several channels.

    

    We see that an end point can support many simultaneous RTP media channels. For each media type, there will typically be one send media channel and one receive media channel; thus, if audio and video are sent in separate RTP streams, there will typically be four media channels. Accompanying the RTP media channels, there is one RTCP media control channel. All of the RTP and RTCP channels run over UDP. In addition to the RTP/RTCP channels, two other channels are required, the call control channel and the call signaling channel. The H.245 call control channel is a TCP connection that carries H.245 control messages.

    Its principal tasks are (i) opening and closing media channels; and (ii) capability exchange, i.e., before sending media, endpoints agree on and encoding algorithm. H.245, being a control protocol for real-time interactive applications, is analogous to RTSP, which is a control protocol for streaming of stored multimedia. Finally, the Q.931 call signaling channel provides classical telephone functionality, such as dial tone and ringing.

    Gatekeepers

    The gatekeeper is an optional H.323 device. Each gatekeeper is responsible for an H.323 zone.

    

    In this deployment scenario, the H.323 terminals and the gatekeeper are all attached to the same LAN, and the H.323 zone is the LAN itself. If a zone has a gatekeeper, then all H.323 terminals in the zone are required to communicate with it using the RAS protocol, which runs over TCP.

    Address translation is one of the more important gatekeeper services. Each terminal can have an alias address, such as the name of the person at the terminal, the e-mail address of the person at the terminal, etc. The gateway translates these alias addresses to IP addresses. This address translation service is similar to the DNS service. Another gatekeeper service is bandwidth management: the gatekeeper can limit the number of simultaneous real-time conferences in order to save some bandwidth for other applications running over the LAN. Optionally, H.323 calls can be routed through gatekeeper, which is useful for billing.

    H.323 terminal must register itself with the gatekeeper in its zone. When the H.323 application is invoked at the terminal, the terminal uses RAS to send its IP address and alias (provided by user) to the gatekeeper. If gatekeeper is present in a zone, each terminal in the zone must contact gatekeeper to ask permission to make a call. Once it has permission, the terminal can send the gatekeeper an e-mail address, alias string or phone extension for the terminal it wants to call, which may be in another zone. If necessary, a gatekeeper will poll other gatekeepers in other zones to resolve an IP address.

    External References:

    • Computer Networking: A Top-Down Approach
    • Request For Comments 1889
    • RFC 3550
    • YouTube Video: Real-time Transport Protocol (RTP) and RTCP | Network Encyclopedia

    https://www.slotsmate.com/learn/slot-machine-rtp

    https://networkencyclopedia.com/real-time-transport-protocol-rtp/