Chapter 2Better meeting and call quality through analytics.
Chapter 2Better meeting and call quality through analytics.
Repeat after me, loud and clear (no pun intended): Call quality matters.
It’s not like you need us to say it. In a recent study1, more than three in four businesses indicated clear audio was the most important factor affecting video conference quality. Call quality routinely disappoints and it can be difficult to identify and resolve the problem. Recent research found that when using video conferencing for work, over a third of respondents indicated they experienced connectivity issues, and almost one in three suffered from audio quality problems2.
In response to the explosion of video meetings and distance learning, companies will need to incorporate high-quality audio and video into their products and services to be successful and meet customer and employee expectations.
This is particularly true for contact centers. Forced to support agents working from home, IT departments are now facing increased responsibility for home internet service quality. It matters: delivering excellent customer experience remains a primary objective for CMOs. See the Marketing Challenge in this blog.
Generally, when a contact center or meetings solution is moved to a cloud provider, you improve agility and simplify operations, but also lose visibility and control over the infrastructure. There is an opportunity to regain that visibility and control for cloud-based contact centers and video solutions using WebRTC technology. New tools that monitor network performance and gauge the customer experience are now available.
Let's first understand what can go wrong.
WebRTC audio quality can suffer for a variety of reasons, including:
- Network performance issues with the underlying transport network, i.e. the agent’s end-to-end connection through the public internet to the contact center as a service platform.
- Software issues with the WebRTC protocol stack running in the agent’s browser, or issues related to signaling, media transmission or NAT traversal.
- Endpoint platform issues like configuration errors, compatibility issues or environmental issues like CPU or memory constraints.
New WebRTC monitoring tools can help you analyze network performance, troubleshoot hardware and software problems, and resolve potential service quality issues.
Repeat after me, loud and clear (no pun intended): Call quality matters.
It’s not like you need us to say it. In a recent study1, more than three in four businesses indicated clear audio was the most important factor affecting video conference quality. Call quality routinely disappoints and it can be difficult to identify and resolve the problem. Recent research found that when using video conferencing for work, over a third of respondents indicated they experienced connectivity issues, and almost one in three suffered from audio quality problems2.
In response to the explosion of video meetings and distance learning, companies will need to incorporate high-quality audio and video into their products and services to be successful and meet customer and employee expectations.
This is particularly true for contact centers. Forced to support agents working from home, IT departments are now facing increased responsibility for home internet service quality. It matters: delivering excellent customer experience remains a primary objective for CMOs. See the Marketing Challenge in this blog.
Generally, when a contact center or meetings solution is moved to a cloud provider, you improve agility and simplify operations, but also lose visibility and control over the infrastructure. There is an opportunity to regain that visibility and control for cloud-based contact centers and video solutions using WebRTC technology. New tools that monitor network performance and gauge the customer experience are now available.
Let's first understand what can go wrong.
WebRTC audio quality can suffer for a variety of reasons, including:
- Network performance issues with the underlying transport network, i.e. the agent’s end-to-end connection through the public internet to the contact center as a service platform.
- Software issues with the WebRTC protocol stack running in the agent’s browser, or issues related to signaling, media transmission or NAT traversal.
- Endpoint platform issues like configuration errors, compatibility issues or environmental issues like CPU or memory constraints.
New WebRTC monitoring tools can help you analyze network performance, troubleshoot hardware and software problems, and resolve potential service quality issues.
Repeat after me, loud and clear (no pun intended): Call quality matters.
It’s not like you need us to say it. In a recent study1, more than three in four businesses indicated clear audio was the most important factor affecting video conference quality. Call quality routinely disappoints and it can be difficult to identify and resolve the problem. Recent research found that when using video conferencing for work, over a third of respondents indicated they experienced connectivity issues, and almost one in three suffered from audio quality problems2.
In response to the explosion of video meetings and distance learning, companies will need to incorporate high-quality audio and video into their products and services to be successful and meet customer and employee expectations.
This is particularly true for contact centers. Forced to support agents working from home, IT departments are now facing increased responsibility for home internet service quality. It matters: delivering excellent customer experience remains a primary objective for CMOs. See the Marketing Challenge in this blog.
Generally, when a contact center or meetings solution is moved to a cloud provider, you improve agility and simplify operations, but also lose visibility and control over the infrastructure. There is an opportunity to regain that visibility and control for cloud-based contact centers and video solutions using WebRTC technology. New tools that monitor network performance and gauge the customer experience are now available.
Let's first understand what can go wrong.
WebRTC audio quality can suffer for a variety of reasons, including:
- Network performance issues with the underlying transport network, i.e. the agent’s end-to-end connection through the public internet to the contact center as a service platform.
- Software issues with the WebRTC protocol stack running in the agent’s browser, or issues related to signaling, media transmission or NAT traversal.
- Endpoint platform issues like configuration errors, compatibility issues or environmental issues like CPU or memory constraints.
New WebRTC monitoring tools can help you analyze network performance, troubleshoot hardware and software problems, and resolve potential service quality issues.
Repeat after me, loud and clear (no pun intended): Call quality matters.
It’s not like you need us to say it. In a recent study1, more than three in four businesses indicated clear audio was the most important factor affecting video conference quality. Call quality routinely disappoints and it can be difficult to identify and resolve the problem. Recent research found that when using video conferencing for work, over a third of respondents indicated they experienced connectivity issues, and almost one in three suffered from audio quality problems2.
In response to the explosion of video meetings and distance learning, companies will need to incorporate high-quality audio and video into their products and services to be successful and meet customer and employee expectations.
This is particularly true for contact centers. Forced to support agents working from home, IT departments are now facing increased responsibility for home internet service quality. It matters: delivering excellent customer experience remains a primary objective for CMOs. See the Marketing Challenge in this blog.
Generally, when a contact center or meetings solution is moved to a cloud provider, you improve agility and simplify operations, but also lose visibility and control over the infrastructure. There is an opportunity to regain that visibility and control for cloud-based contact centers and video solutions using WebRTC technology. New tools that monitor network performance and gauge the customer experience are now available.
Let's first understand what can go wrong.
WebRTC audio quality can suffer for a variety of reasons, including:
- Network performance issues with the underlying transport network, i.e. the agent’s end-to-end connection through the public internet to the contact center as a service platform.
- Software issues with the WebRTC protocol stack running in the agent’s browser, or issues related to signaling, media transmission or NAT traversal.
- Endpoint platform issues like configuration errors, compatibility issues or environmental issues like CPU or memory constraints.
New WebRTC monitoring tools can help you analyze network performance, troubleshoot hardware and software problems, and resolve potential service quality issues.
Why monitor endpoints?
Endpoints, such as contact center agents, provide the best vantage point to gauge the user experience. As WebRTC is implemented in the browser, the best place to get WebRTC performance data is directly from the browser running on a WebRTC endpoint. The WebRTC getStats API supports an extensive collection of real-time communications statistics that can be accessed directly from an agent’s browser.
Generally speaking, the WebRTC getStats API supports three types of endpoint statistics that are vital to analyzing performance and troubleshooting problems. They correspond to stages in the media pipeline, as described in Varun Singh and Marcin Nagy’s blog How Problems in the WebRTC Media Pipeline Affect Quality of Experience:
- Network connectivity - packet transport metrics, including throughput, loss, delay and jitter
- Media - media stream metrics, including throughput per channel (audio, video and data)
- Signaling - metrics for session negotiation (SDP or XMPP) and address resolution (STUN/TURN/ICE).
Typical challenges associated with using the getStats API to monitor WebRTC endpoints include:
- Keeping track of the WebRTC statistics each browser vendor supports
- Ensuring endpoint statistics are collected and transported efficiently and securely
Why monitor endpoints?
Endpoints, such as contact center agents, provide the best vantage point to gauge the user experience. As WebRTC is implemented in the browser, the best place to get WebRTC performance data is directly from the browser running on a WebRTC endpoint. The WebRTC getStats API supports an extensive collection of real-time communications statistics that can be accessed directly from an agent’s browser.
Generally speaking, the WebRTC getStats API supports three types of endpoint statistics that are vital to analyzing performance and troubleshooting problems. They correspond to stages in the media pipeline, as described in Varun Singh and Marcin Nagy’s blog How Problems in the WebRTC Media Pipeline Affect Quality of Experience:
- Network connectivity - packet transport metrics, including throughput, loss, delay and jitter
- Media - media stream metrics, including throughput per channel (audio, video and data)
- Signaling - metrics for session negotiation (SDP or XMPP) and address resolution (STUN/TURN/ICE).
Typical challenges associated with using the getStats API to monitor WebRTC endpoints include:
- Keeping track of the WebRTC statistics each browser vendor supports
- Ensuring endpoint statistics are collected and transported efficiently and securely
Why monitor endpoints?
Endpoints, such as contact center agents, provide the best vantage point to gauge the user experience. As WebRTC is implemented in the browser, the best place to get WebRTC performance data is directly from the browser running on a WebRTC endpoint. The WebRTC getStats API supports an extensive collection of real-time communications statistics that can be accessed directly from an agent’s browser.
Generally speaking, the WebRTC getStats API supports three types of endpoint statistics that are vital to analyzing performance and troubleshooting problems. They correspond to stages in the media pipeline, as described in Varun Singh and Marcin Nagy’s blog How Problems in the WebRTC Media Pipeline Affect Quality of Experience:
- Network connectivity - packet transport metrics, including throughput, loss, delay and jitter
- Media - media stream metrics, including throughput per channel (audio, video and data)
- Signaling - metrics for session negotiation (SDP or XMPP) and address resolution (STUN/TURN/ICE).
Typical challenges associated with using the getStats API to monitor WebRTC endpoints include:
- Keeping track of the WebRTC statistics each browser vendor supports
- Ensuring endpoint statistics are collected and transported efficiently and securely
Why monitor endpoints?
Endpoints, such as contact center agents, provide the best vantage point to gauge the user experience. As WebRTC is implemented in the browser, the best place to get WebRTC performance data is directly from the browser running on a WebRTC endpoint. The WebRTC getStats API supports an extensive collection of real-time communications statistics that can be accessed directly from an agent’s browser.
Generally speaking, the WebRTC getStats API supports three types of endpoint statistics that are vital to analyzing performance and troubleshooting problems. They correspond to stages in the media pipeline, as described in Varun Singh and Marcin Nagy’s blog How Problems in the WebRTC Media Pipeline Affect Quality of Experience:
- Network connectivity - packet transport metrics, including throughput, loss, delay and jitter
- Media - media stream metrics, including throughput per channel (audio, video and data)
- Signaling - metrics for session negotiation (SDP or XMPP) and address resolution (STUN/TURN/ICE).
Typical challenges associated with using the getStats API to monitor WebRTC endpoints include:
- Keeping track of the WebRTC statistics each browser vendor supports
- Ensuring endpoint statistics are collected and transported efficiently and securely
Tracking browser releases is a headache.
The W3C/IETF getStats API specifications outline hundreds of potential WebRTC statistics. The vast majority of these statistics are optional, so support varies widely between browsers and browser release versions.
To further complicate matters, browser releases happen frequently. Google, for example, introduces new major releases of Chrome every six weeks, sometimes adding new stats. In any given contact center or company using video conferencing, a variety of browsers may be in use (Chrome, Firefox, Safari, etc.) at different release levels, so it’s important for a monitoring software provider to follow browser releases closely and perform regular regression tests to avoid incompatibilities and keep pace with change.
Tracking browser releases is a headache.
The W3C/IETF getStats API specifications outline hundreds of potential WebRTC statistics. The vast majority of these statistics are optional, so support varies widely between browsers and browser release versions.
To further complicate matters, browser releases happen frequently. Google, for example, introduces new major releases of Chrome every six weeks, sometimes adding new stats. In any given contact center or company using video conferencing, a variety of browsers may be in use (Chrome, Firefox, Safari, etc.) at different release levels, so it’s important for a monitoring software provider to follow browser releases closely and perform regular regression tests to avoid incompatibilities and keep pace with change.
Tracking browser releases is a headache.
The W3C/IETF getStats API specifications outline hundreds of potential WebRTC statistics. The vast majority of these statistics are optional, so support varies widely between browsers and browser release versions.
To further complicate matters, browser releases happen frequently. Google, for example, introduces new major releases of Chrome every six weeks, sometimes adding new stats. In any given contact center or company using video conferencing, a variety of browsers may be in use (Chrome, Firefox, Safari, etc.) at different release levels, so it’s important for a monitoring software provider to follow browser releases closely and perform regular regression tests to avoid incompatibilities and keep pace with change.
Tracking browser releases is a headache.
The W3C/IETF getStats API specifications outline hundreds of potential WebRTC statistics. The vast majority of these statistics are optional, so support varies widely between browsers and browser release versions.
To further complicate matters, browser releases happen frequently. Google, for example, introduces new major releases of Chrome every six weeks, sometimes adding new stats. In any given contact center or company using video conferencing, a variety of browsers may be in use (Chrome, Firefox, Safari, etc.) at different release levels, so it’s important for a monitoring software provider to follow browser releases closely and perform regular regression tests to avoid incompatibilities and keep pace with change.
Herding the “statistics cats” into one place.
WebRTC statistics must be transported from endpoints to a central repository for consolidation, analysis and reporting. This often requires a tradeoff between the breadth and depth of the data collected and the network bandwidth consumed. On one hand, you want to capture and forward as much statistical data as possible, as frequently as possible for ultimate granularity. On the other hand, you don’t want to overwhelm the network with statistical data. (In extreme cases, WebRTC performance monitoring and troubleshooting tools can actually impair service quality and exacerbate problems by seizing bandwidth, a phenomenon known as the observer effect in physics.) Unfortunately, the less frequently you capture statistics, the more likely you are to miss a short-lived event like burst packet loss.
Herding the “statistics cats” into one place.
WebRTC statistics must be transported from endpoints to a central repository for consolidation, analysis and reporting. This often requires a tradeoff between the breadth and depth of the data collected and the network bandwidth consumed. On one hand, you want to capture and forward as much statistical data as possible, as frequently as possible for ultimate granularity. On the other hand, you don’t want to overwhelm the network with statistical data. (In extreme cases, WebRTC performance monitoring and troubleshooting tools can actually impair service quality and exacerbate problems by seizing bandwidth, a phenomenon known as the observer effect in physics.) Unfortunately, the less frequently you capture statistics, the more likely you are to miss a short-lived event like burst packet loss.
Herding the “statistics cats” into one place.
WebRTC statistics must be transported from endpoints to a central repository for consolidation, analysis and reporting. This often requires a tradeoff between the breadth and depth of the data collected and the network bandwidth consumed. On one hand, you want to capture and forward as much statistical data as possible, as frequently as possible for ultimate granularity. On the other hand, you don’t want to overwhelm the network with statistical data. (In extreme cases, WebRTC performance monitoring and troubleshooting tools can actually impair service quality and exacerbate problems by seizing bandwidth, a phenomenon known as the observer effect in physics.) Unfortunately, the less frequently you capture statistics, the more likely you are to miss a short-lived event like burst packet loss.
Herding the “statistics cats” into one place.
WebRTC statistics must be transported from endpoints to a central repository for consolidation, analysis and reporting. This often requires a tradeoff between the breadth and depth of the data collected and the network bandwidth consumed. On one hand, you want to capture and forward as much statistical data as possible, as frequently as possible for ultimate granularity. On the other hand, you don’t want to overwhelm the network with statistical data. (In extreme cases, WebRTC performance monitoring and troubleshooting tools can actually impair service quality and exacerbate problems by seizing bandwidth, a phenomenon known as the observer effect in physics.) Unfortunately, the less frequently you capture statistics, the more likely you are to miss a short-lived event like burst packet loss.
Augmenting getStats API data.
Developing a complete view of the user experience requires collecting and examining statistical data from both endpoints in a WebRTC session. Most cloud contact centers establish simple point-to-point WebRTC connections between agents and a PSTN gateway in the contact center as a service infrastructure. Unfortunately, the PSTN gateway in the CCaaS infrastructure typically does not support the getStats API. You can use the Real-time Transport Control Protocol (RTCP) as an alternative to the getStats API to gain visibility into the PSTN gateway side of the session.
RTCP allows one endpoint to exchange performance statistics with other endpoints in a WebRTC session. RTCP statistics are robust and include delay, loss, jitter and throughput measurements (plus many more statistics if RFC 3611 RTCP Extended Reports [XR] are enabled). By combining getStats API data from an agent endpoint with RTCP data you can obtain a full, end-to-end view of the session.
You can also use RTCP to gather statistics from intermediary network elements like MCUs and SFUs in WebRTC conferencing and collaboration applications. By monitoring both endpoints of the session, along with intermediary network elements you can increase the depth and breadth of the statistics you gather, which can help you identify and resolve issues more quickly and efficiently.
Augmenting getStats API data.
Developing a complete view of the user experience requires collecting and examining statistical data from both endpoints in a WebRTC session. Most cloud contact centers establish simple point-to-point WebRTC connections between agents and a PSTN gateway in the contact center as a service infrastructure. Unfortunately, the PSTN gateway in the CCaaS infrastructure typically does not support the getStats API. You can use the Real-time Transport Control Protocol (RTCP) as an alternative to the getStats API to gain visibility into the PSTN gateway side of the session.
RTCP allows one endpoint to exchange performance statistics with other endpoints in a WebRTC session. RTCP statistics are robust and include delay, loss, jitter and throughput measurements (plus many more statistics if RFC 3611 RTCP Extended Reports [XR] are enabled). By combining getStats API data from an agent endpoint with RTCP data you can obtain a full, end-to-end view of the session.
You can also use RTCP to gather statistics from intermediary network elements like MCUs and SFUs in WebRTC conferencing and collaboration applications. By monitoring both endpoints of the session, along with intermediary network elements you can increase the depth and breadth of the statistics you gather, which can help you identify and resolve issues more quickly and efficiently.
Augmenting getStats API data.
Developing a complete view of the user experience requires collecting and examining statistical data from both endpoints in a WebRTC session. Most cloud contact centers establish simple point-to-point WebRTC connections between agents and a PSTN gateway in the contact center as a service infrastructure. Unfortunately, the PSTN gateway in the CCaaS infrastructure typically does not support the getStats API. You can use the Real-time Transport Control Protocol (RTCP) as an alternative to the getStats API to gain visibility into the PSTN gateway side of the session.
RTCP allows one endpoint to exchange performance statistics with other endpoints in a WebRTC session. RTCP statistics are robust and include delay, loss, jitter and throughput measurements (plus many more statistics if RFC 3611 RTCP Extended Reports [XR] are enabled). By combining getStats API data from an agent endpoint with RTCP data you can obtain a full, end-to-end view of the session.
You can also use RTCP to gather statistics from intermediary network elements like MCUs and SFUs in WebRTC conferencing and collaboration applications. By monitoring both endpoints of the session, along with intermediary network elements you can increase the depth and breadth of the statistics you gather, which can help you identify and resolve issues more quickly and efficiently.
Augmenting getStats API data.
Developing a complete view of the user experience requires collecting and examining statistical data from both endpoints in a WebRTC session. Most cloud contact centers establish simple point-to-point WebRTC connections between agents and a PSTN gateway in the contact center as a service infrastructure. Unfortunately, the PSTN gateway in the CCaaS infrastructure typically does not support the getStats API. You can use the Real-time Transport Control Protocol (RTCP) as an alternative to the getStats API to gain visibility into the PSTN gateway side of the session.
RTCP allows one endpoint to exchange performance statistics with other endpoints in a WebRTC session. RTCP statistics are robust and include delay, loss, jitter and throughput measurements (plus many more statistics if RFC 3611 RTCP Extended Reports [XR] are enabled). By combining getStats API data from an agent endpoint with RTCP data you can obtain a full, end-to-end view of the session.
You can also use RTCP to gather statistics from intermediary network elements like MCUs and SFUs in WebRTC conferencing and collaboration applications. By monitoring both endpoints of the session, along with intermediary network elements you can increase the depth and breadth of the statistics you gather, which can help you identify and resolve issues more quickly and efficiently.
In a peer-to-peer WebRTC application like a video chat you can get a complete view of a session by monitoring both endpoints using the getStats API, or by monitoring one endpoint via the getStats API and the other via RCTP stats.
In a peer-to-peer WebRTC application like a video chat you can get a complete view of a session by monitoring both endpoints using the getStats API, or by monitoring one endpoint via the getStats API and the other via RCTP stats.
In a peer-to-peer WebRTC application like a video chat you can get a complete view of a session by monitoring both endpoints using the getStats API, or by monitoring one endpoint via the getStats API and the other via RCTP stats.
In a peer-to-peer WebRTC application like a video chat you can get a complete view of a session by monitoring both endpoints using the getStats API, or by monitoring one endpoint via the getStats API and the other via RCTP stats.
World-class quality made easy with callstats.
8x8 callstats is built to optimize WebRTC audio quality and improve user experiences. It embeds advanced monitoring functionality into WebRTC endpoints, giving operations teams real-time visibility into key network performance indicators and service quality metrics. The solution gathers all supported WebRTC statistics from each endpoint, transforming raw data into actionable insights.
It used an adaptive querying/reporting algorithm to balance statistical granularity with bandwidth consumption. At the beginning of a WebRTC session the callstats client queries the browser for statistics every second, and sends the results to an upstream data collector. After filtering out anomalies, the querying/reporting frequency is reduced to conserve bandwidth. The adaptive stats algorithm provides full visibility into key performance metrics, without overburdening the network or impairing the user experience.
Of course, callstats employs strong security measures to protect the privacy of WebRTC metadata. We authenticate callstats clients to prevent masquerading, encrypt data in transit to prevent eavesdropping and man-in-the-middle attacks, encrypt data at rest to protect data confidentiality, and implement strong access control mechanisms to prevent unauthorized data disclosure.
Since WebRTC is the sole focus, browser releases are closely monitored with continuous regression testing against the latest releases. 8x8 also maintains a close working relationship with Google so we can closely track the latest Chrome developments.
World-class quality made easy with callstats.
8x8 callstats is built to optimize WebRTC audio quality and improve user experiences. It embeds advanced monitoring functionality into WebRTC endpoints, giving operations teams real-time visibility into key network performance indicators and service quality metrics. The solution gathers all supported WebRTC statistics from each endpoint, transforming raw data into actionable insights.
It used an adaptive querying/reporting algorithm to balance statistical granularity with bandwidth consumption. At the beginning of a WebRTC session the callstats client queries the browser for statistics every second, and sends the results to an upstream data collector. After filtering out anomalies, the querying/reporting frequency is reduced to conserve bandwidth. The adaptive stats algorithm provides full visibility into key performance metrics, without overburdening the network or impairing the user experience.
Of course, callstats employs strong security measures to protect the privacy of WebRTC metadata. We authenticate callstats clients to prevent masquerading, encrypt data in transit to prevent eavesdropping and man-in-the-middle attacks, encrypt data at rest to protect data confidentiality, and implement strong access control mechanisms to prevent unauthorized data disclosure.
Since WebRTC is the sole focus, browser releases are closely monitored with continuous regression testing against the latest releases. 8x8 also maintains a close working relationship with Google so we can closely track the latest Chrome developments.
World-class quality made easy with callstats.
8x8 callstats is built to optimize WebRTC audio quality and improve user experiences. It embeds advanced monitoring functionality into WebRTC endpoints, giving operations teams real-time visibility into key network performance indicators and service quality metrics. The solution gathers all supported WebRTC statistics from each endpoint, transforming raw data into actionable insights.
It used an adaptive querying/reporting algorithm to balance statistical granularity with bandwidth consumption. At the beginning of a WebRTC session the callstats client queries the browser for statistics every second, and sends the results to an upstream data collector. After filtering out anomalies, the querying/reporting frequency is reduced to conserve bandwidth. The adaptive stats algorithm provides full visibility into key performance metrics, without overburdening the network or impairing the user experience.
Of course, callstats employs strong security measures to protect the privacy of WebRTC metadata. We authenticate callstats clients to prevent masquerading, encrypt data in transit to prevent eavesdropping and man-in-the-middle attacks, encrypt data at rest to protect data confidentiality, and implement strong access control mechanisms to prevent unauthorized data disclosure.
Since WebRTC is the sole focus, browser releases are closely monitored with continuous regression testing against the latest releases. 8x8 also maintains a close working relationship with Google so we can closely track the latest Chrome developments.
World-class quality made easy with callstats.
8x8 callstats is built to optimize WebRTC audio quality and improve user experiences. It embeds advanced monitoring functionality into WebRTC endpoints, giving operations teams real-time visibility into key network performance indicators and service quality metrics. The solution gathers all supported WebRTC statistics from each endpoint, transforming raw data into actionable insights.
It used an adaptive querying/reporting algorithm to balance statistical granularity with bandwidth consumption. At the beginning of a WebRTC session the callstats client queries the browser for statistics every second, and sends the results to an upstream data collector. After filtering out anomalies, the querying/reporting frequency is reduced to conserve bandwidth. The adaptive stats algorithm provides full visibility into key performance metrics, without overburdening the network or impairing the user experience.
Of course, callstats employs strong security measures to protect the privacy of WebRTC metadata. We authenticate callstats clients to prevent masquerading, encrypt data in transit to prevent eavesdropping and man-in-the-middle attacks, encrypt data at rest to protect data confidentiality, and implement strong access control mechanisms to prevent unauthorized data disclosure.
Since WebRTC is the sole focus, browser releases are closely monitored with continuous regression testing against the latest releases. 8x8 also maintains a close working relationship with Google so we can closely track the latest Chrome developments.
Chapter Two Takeaways
Deploying a cloud-based contact center or meetings solution requires new tools to monitor WebRTC performance and service quality. To successfully deliver the required visibility and control to the experience, it’s important to consider the performance implications of collecting and analyzing massive WebRTC datasets. 8x8 callstats makes it all easy with a secure, real-time solution that puts control of the customer experience in your hands - even for cloud-based (WebRTC) contact center and meetings applications.
Check out this product tour that walks you through how callstats works and see how easy it is to use analytics that provide visibility and control for cloud-based contact center and meeting applications.
Helpful Resources
Video: callstats from Amazon Connect
Chapter Two Takeaways
Deploying a cloud-based contact center or meetings solution requires new tools to monitor WebRTC performance and service quality. To successfully deliver the required visibility and control to the experience, it’s important to consider the performance implications of collecting and analyzing massive WebRTC datasets. 8x8 callstats makes it all easy with a secure, real-time solution that puts control of the customer experience in your hands - even for cloud-based (WebRTC) contact center and meetings applications.
Check out this product tour that walks you through how callstats works and see how easy it is to use analytics that provide visibility and control for cloud-based contact center and meeting applications.
Helpful Resources
Video: callstats from Amazon Connect
Chapter Two Takeaways
Deploying a cloud-based contact center or meetings solution requires new tools to monitor WebRTC performance and service quality. To successfully deliver the required visibility and control to the experience, it’s important to consider the performance implications of collecting and analyzing massive WebRTC datasets. 8x8 callstats makes it all easy with a secure, real-time solution that puts control of the customer experience in your hands - even for cloud-based (WebRTC) contact center and meetings applications.
Check out this product tour that walks you through how callstats works and see how easy it is to use analytics that provide visibility and control for cloud-based contact center and meeting applications.
Helpful Resources
Video: callstats from Amazon Connect
Chapter Two Takeaways
Deploying a cloud-based contact center or meetings solution requires new tools to monitor WebRTC performance and service quality. To successfully deliver the required visibility and control to the experience, it’s important to consider the performance implications of collecting and analyzing massive WebRTC datasets. 8x8 callstats makes it all easy with a secure, real-time solution that puts control of the customer experience in your hands - even for cloud-based (WebRTC) contact center and meetings applications.
Check out this product tour that walks you through how callstats works and see how easy it is to use analytics that provide visibility and control for cloud-based contact center and meeting applications.
Helpful Resources
Video: callstats from Amazon Connect
Chapter Three Preview
Our buying behaviors have changed, and quickly. The speed and degree of change in your and my buying behaviors has many marketing organizations scrambling to remodel their approaches. One of the keys to success is how fast marketing departments are able to adjust. In the next chapter of our eBook, Build Your Experience, we will share 7 SMS marketing tips that will give you the tools to meet those rapidly changing behaviors head on with 160 effective characters.
Chapter Three Preview
Our buying behaviors have changed, and quickly. The speed and degree of change in your and my buying behaviors has many marketing organizations scrambling to remodel their approaches. One of the keys to success is how fast marketing departments are able to adjust. In the next chapter of our eBook, Build Your Experience, we will share 7 SMS marketing tips that will give you the tools to meet those rapidly changing behaviors head on with 160 effective characters.
Chapter Three Preview
Our buying behaviors have changed, and quickly. The speed and degree of change in your and my buying behaviors has many marketing organizations scrambling to remodel their approaches. One of the keys to success is how fast marketing departments are able to adjust. In the next chapter of our eBook, Build Your Experience, we will share 7 SMS marketing tips that will give you the tools to meet those rapidly changing behaviors head on with 160 effective characters.
Chapter Three Preview
Our buying behaviors have changed, and quickly. The speed and degree of change in your and my buying behaviors has many marketing organizations scrambling to remodel their approaches. One of the keys to success is how fast marketing departments are able to adjust. In the next chapter of our eBook, Build Your Experience, we will share 7 SMS marketing tips that will give you the tools to meet those rapidly changing behaviors head on with 160 effective characters.
8x8 is trusted by over three million users worldwide.
8x8 is trusted by over three million users worldwide.
8x8 is trusted by over three million users worldwide.
8x8 is trusted by over three million users worldwide.
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