Updated small changes

Author: sloanster

articles/communication-services/concepts/voice-video-calling/calling-sdk-features.md

@@ -7,15 +7,15 @@ manager: chpalm
 services: azure-communication-services
 
 ms.author: chpalm
-ms.date: 06/30/2021
+ms.date: 02/24/2024
 ms.topic: conceptual
 ms.service: azure-communication-services
 ms.subservice: calling
 ms.custom: devx-track-js
 ---
 # Calling SDK overview
 
-Azure Communication Services allows end-user browsers, apps, and services to drive voice and video communication. This page focuses on Calling client SDK, which can be embedded in websites and native applications. This page provides detailed descriptions of Calling client features such as platform and browser support information. Services programmatically manage and access calls using the [Call Automation APIs](../call-automation/call-automation.md). The [Rooms API](../rooms/room-concept.md) is an optional Azure Communication Services API that adds additional features to a voice or video call, such as roles and permissions.
+Azure Communication Services allows end-user browsers, apps, and services to drive voice and video communication. This page focuses on Calling client SDK, which can be embedded in websites and native applications. This page provides detailed descriptions of Calling client features such as platform and browser support information. Services programmatically manages and access calls using the [Call Automation APIs](../call-automation/call-automation.md). The [Rooms API](../rooms/room-concept.md) is an optional Azure Communication Services API that adds additional features to a voice or video call, such as roles and permissions.
 
 [!INCLUDE [Survey Request](../../includes/survey-request.md)]
 
@@ -166,7 +166,7 @@ The Azure Communication Services Calling SDK supports the following streaming co
 | **Maximum # of incoming remote streams that can be rendered simultaneously** | 9 videos + 1 screen sharing on desktop browsers*, 4 videos + 1 screen sharing on web mobile browsers | 9 videos + 1 screen sharing |
 
 \* Starting from Azure Communication Services Web Calling SDK version [1.16.3](https://github.com/Azure/Communication/blob/master/releasenotes/acs-javascript-calling-library-release-notes.md#1163-stable-2023-08-24)
-While the Calling SDK does not enforce these limits, your users might experience performance degradation if they're exceeded. Use the API of [Optimal Video Count](../../how-tos/calling-sdk/manage-video.md?pivots=platform-web#remote-video-quality) to determine how many current incoming video streams your web environment can support.
+While the Calling SDK doesn't enforce these limits, your users might experience performance degradation if they're exceeded. Use the API of [Optimal Video Count](../../how-tos/calling-sdk/manage-video.md?pivots=platform-web#remote-video-quality) to determine how many current incoming video streams your web environment can support.
 
 ## Supported video resolutions
 The Azure Communicaton Services Calling SDK support the following video resolutions:

articles/communication-services/concepts/voice-video-calling/network-requirements.md

@@ -7,7 +7,7 @@ manager: chpalm
 services: azure-communication-services
 
 ms.author: nmurav
-ms.date: 09/12/2023
+ms.date: 02/19/2024
 ms.topic: conceptual
 ms.service: azure-communication-services
 ms.subservice: calling
@@ -24,7 +24,7 @@ The quality of real-time media over IP is significantly affected by the quality
 
 * **Latency**. The time it takes to get an IP packet from point A to point B on the network. This network propagation delay is determined by the physical distance between the two points and any other overhead incurred by the devices that your traffic flows through. Latency is measured as one-way or round-trip time (RTT).
 * **Packet loss**. A percentage of packets that are lost in a specific window of time. Packet loss directly affects audio quality—from small, individual lost packets having almost no impact to back-to-back burst losses that cause complete audio cut-out.
-* **Inter-packet arrival jitter, also known as jitter**. The average change in delay between successive packets. Communication Services can adapt to some levels of jitter through buffering. It's only when the jitter exceeds the buffering that a participant will notice its effects.
+* **Inter-packet arrival jitter, also known as jitter**. The average change in delay between successive packets. Communication Services can adapt to some levels of jitter through buffering. It's only when the jitter exceeds the buffering that a participant notices its effects.
 
 ## Network bandwidth
 
@@ -56,15 +56,15 @@ The following bandwidth requirements are for the native Windows, Android, and iO
 
 ## Firewall configuration
 
-Communication Services connections require internet connectivity to specific ports and IP addresses to deliver high-quality multimedia experiences. Without access to these ports and IP addresses, Communication Services will not work properly. The list of IP ranges and allow listed domains that need to be enabled are:
+Communication Services connections require internet connectivity to specific ports and IP addresses to deliver high-quality multimedia experiences. Without access to these ports and IP addresses, Communication Services won't work properly. The list of IP ranges and allow listed domains that need to be enabled are:
 
 | Category | IP ranges or FQDN | Ports | 
 | :-- | :-- | :-- |
 | Media traffic | Range of Azure public cloud IP addresses 20.202.0.0/16 The range provided above is the range of IP addresses on either Media processor or Azure Communication Services TURN service. | UDP 3478 through 3481, TCP ports 443 |
 | Signaling, telemetry, registration| *.skype.com, *.microsoft.com, *.azure.net, *.azure.com, *.office.com| TCP 443, 80 |
 
 
-The endpoints below should be reachable for U.S. Government GCC High customers only
+The endpoints below should be reachable for U.S. Government GCC High customers only.
 
 | Category | IP ranges or FQDN | Ports | 
 | :-- | :-- | :-- |

articles/communication-services/concepts/voice-video-calling/simulcast.md

@@ -1,7 +1,7 @@
 ---
 title: Azure Communication Services Simulcast
 titleSuffix: An Azure Communication Services concept document
-description: Overview of Simulcast - how sending multiple video quality streams helps overall call quality
+description: Overview of Simulcast - how sending multiple video quality streams helps overall call quality.
 author: sloanster
 manager: chpalm
 services: azure-communication-services
@@ -32,4 +32,4 @@ When streaming with simulcast, there are no set resolutions for high or low qual
 - 540p
 - 360p
 - 240p
-- 180p
+- 180p