Release Notes & Changelogs

• Fixed an issue where a black flicker could briefly appear on a subscribed stream immediately after subscribing during 1-on-1 calls on Safari. • Fixed an issue where retrieving WebRTC statistics could throw a TypeError on iOS WKWebView and Capacitor-wrapped apps, causing errors when collecting stats or running network tests. • Fixed an issue on Firefox where rapidly toggling video off and on (calling publishVideo(false) immediately followed by publishVideo(true)) could intermittently leave the published video disabled, showing a black frame until the next toggle. • Fixed an issue where the audioLocalCandidateType, videoLocalCandidateType, audioRemoteCandidateType, and videoRemoteCandidateType fields in the QoS stats could report incorrect values by using the last ICE candidate instead of the selected one. • Fixed an issue where a subscriber could fail to start when the browser's autoplay policy blocked video playback. The subscriber now times out the play attempt and continues with muted audio (which the user can unmute) instead of failing to subscribe.

• Fixed an issue where a black flicker could briefly appear on a subscribed stream immediately after subscribing during 1-on-1 calls on Safari. • Fixed an issue where retrieving WebRTC statistics could throw a TypeError on iOS WKWebView and Capacitor-wrapped apps, causing errors when collecting stats or running network tests. • Fixed an issue on Firefox where rapidly toggling video off and on (calling publishVideo(false) immediately followed by publishVideo(true)) could intermittently leave the published video disabled, showing a black frame until the next toggle. • Fixed an issue where the audioLocalCandidateType, videoLocalCandidateType, audioRemoteCandidateType, and videoRemoteCandidateType fields in the QoS stats could report incorrect values by using the last ICE candidate instead of the selected one. • Fixed an issue where a subscriber could fail to start when the browser's autoplay policy blocked video playback. The subscriber now times out the play attempt and continues with muted audio (which the user can unmute) instead of failing to subscribe.

• OpenTok.js release notes. • Updated to use version 5.1.6 of the Vonage Video API Network Test JS library. • Live Captions metrics in Session and Session Summary. Two new fields, hasCaptions and liveCaptionsPublisherMinutes, are now available on the Session and SessionSummary types, enabling you to query Live Captions usage across your sessions. • Audio Connector minutes in Session and Session Summary. The audioConnectorMinutes field is now available on both the Session and SessionSummary types, reporting the total duration (in minutes) that an Audio Connector was connected during a session. This enables more detailed billing and usage reporting for Audio Connector integrations. • Post-Call Transcription minutes in usage data. The postCallTranscriptionMinutes field is now available on the Usage type, under both the applicationData and projectData schemas, reporting the total number of post-call transcription minutes.

• Account Takeover Protection. Protect your Vonage APIs from unauthorized access with IP Allowlisting. Add trusted IP addresses or IP ranges to your allowlist and only those addresses will be able to call your Vonage APIs. Any request from an unlisted address is automatically blocked, keeping your account protected even if credentials are ever compromised. Check our developer guide to know more.

• OpenTok.js release notes. • Updated to use version 5.1.6 of the Network Test JS library. • Live Captions metrics in Session and Session Summary. Two new fields, hasCaptions and liveCaptionsPublisherMinutes, are now available on the Session and SessionSummary types, enabling you to query Live Captions usage across your sessions. • Audio Connector minutes in Session and Session Summary. The audioConnectorMinutes field is now available on both the Session and SessionSummary types, reporting the total duration (in minutes) that an Audio Connector was connected during a session. This enables more detailed billing and usage reporting for Audio Connector integrations. • Post-Call Transcription minutes in usage data. The postCallTranscriptionMinutes field is now available on the Usage type, under both the applicationData and projectData schemas, reporting the total number of post-call transcription minutes.

• Promise support for async SDK methods (non-breaking). The JS SDK now supports modern async/await and promise-based patterns alongside the existing callback-based API. This change is fully backward compatible — all existing callback-based code continues to work without modification. As part of this work, the SDK gained finer control over errors in its internal flows and now surfaces error information more accurately. Starting with version 2.35.1, some functions report errors that were previously silent. These are not new failures — they are pre-existing errors that are now correctly propagated, so you may observe errors being reported in cases that previously went unnoticed. For more information, see this documentation. • Scalability Mode API. This version adds a new API that allows publishers to explicitly set the target scalability mode, giving developers fine-grained control over the number of spatial (resolution) and temporal (frame rate) layers used during encoding. This enables applications to better balance video quality, adaptability, and resource usage based on their specific needs. For more information, see this documentation. This is a beta feature. • Performed Android 17 validation. The Vonage Web SDK has been validated on Android 17, confirming smooth operation across Android mobile browsers and WebView applications. • Fixed an issue where, in some cases, subscribing to a VP9 stream failed when the publisher had set videoContentHint to "detail" or "text". • Fixed a memory leak in audio processing across subscribe/unsubscribe and publish/unpublish cycles that in some cases could lead to unbounded heap growth and audio degradation over time.

• This version adds a new API that allows publishers to explicitly set the target scalability mode, giving developers fine-grained control over the number of spatial (resolution) and temporal (frame rate) layers used during encoding. This enables applications to better balance video quality, adaptability, and resource usage based on their specific needs. For more information, see this documentation. This is a beta feature. • This version of the SDK now uses WebRTC M145, a recent milestone from Google's WebRTC library. The upgrade brings improvements in performance, stability, and security across all supported native platforms. It includes bug fixes and performance enhancements to audio and video media codecs, better network adaptation, and improved compatibility with modern platform environments. This update ensures the SDK remains aligned with the current WebRTC specification and maintains a strong foundation for future feature development. • This version includes an update to the bundled CA certificates used by the SDK for establishing secure connections. This file contains trusted Certificate Authority certificates that the SDK uses to verify SSL/TLS connections, ensuring secure communication with external services. This file serves as a fallback and is only used when the SDK cannot locate a CA certificate bundle in the system. This routine update adds newly trusted CAs and removes outdated certificates. • The otc_video_frame_set_metadata() and otc_video_frame_get_metadata() functions, deprecated in 2.33.0, have been removed in this version. • Fixed an issue where calling otc_session_publish() inside the on_session_connected callback could fail due to a race condition caused by the session's internal connected state not being updated before the callback fired.

• This version adds a new API that allows publishers to explicitly set the target scalability mode, giving developers fine-grained control over the number of spatial (resolution) and temporal (frame rate) layers used during encoding. This enables applications to better balance video quality, adaptability, and resource usage based on their specific needs. For more information, see this documentation. This is a beta feature. • This version of the SDK now uses WebRTC M145, a recent milestone from Google's WebRTC library. The upgrade brings improvements in performance, stability, and security across all supported native platforms. It includes bug fixes and performance enhancements to audio and video media codecs, better network adaptation, and improved compatibility with modern platform environments. This update ensures the SDK remains aligned with the current WebRTC specification and maintains a strong foundation for future feature development. • The Metadata property of VideoFrame, deprecated in 2.33.0, has been removed in this version. • Setting a non-default target scalability mode may affect the behaviour of setPreferredResolution() and setPreferredFrameRate() for subscribers. For more information, see the Interaction with subscriber preferred resolution and frame rate note.

• This version adds a new API that allows publishers to explicitly set the target scalability mode, giving developers fine-grained control over the number of spatial (resolution) and temporal (frame rate) layers used during encoding. This enables applications to better balance video quality, adaptability, and resource usage based on their specific needs. For more information, see this documentation. This is a beta feature. • This version of the SDK now uses WebRTC M145, a recent milestone from Google's WebRTC library. The upgrade brings improvements in performance, stability, and security across all supported native platforms. It includes bug fixes and performance enhancements to audio and video media codecs, better network adaptation, and improved compatibility with modern platform environments. This update ensures the SDK remains aligned with the current WebRTC specification and maintains a strong foundation for future feature development. • The Metadata property of VideoFrame, deprecated in 2.33.0, has been removed in this version. • Setting a non-default target scalability mode may affect the behaviour of setPreferredResolution() and setPreferredFrameRate() for subscribers. For more information, see the Interaction with subscriber preferred resolution and frame rate note.

• This version adds a new API that allows publishers to explicitly set the target scalability mode, giving developers fine-grained control over the number of spatial (resolution) and temporal (frame rate) layers used during encoding. This enables applications to better balance video quality, adaptability, and resource usage based on their specific needs. For more information, see this documentation. This is a beta feature. • This version of the SDK now uses WebRTC M145, a recent milestone from Google's WebRTC library. The upgrade brings improvements in performance, stability, and security across all supported native platforms. It includes bug fixes and performance enhancements to audio and video media codecs, better network adaptation, and improved compatibility with modern platform environments. This update ensures the SDK remains aligned with the current WebRTC specification and maintains a strong foundation for future feature development. • Added a new Installation section to the iOS SDK overview documentation covering: • Swift Package Manager (SPM) as the recommended installation method (available since v2.35.1). • CocoaPods (OTXCFramework pod) as a maintained but deprecated alternative.