Try new features by downloading our latest update of WebRTC SIP SDK for web. Enhanced overall stability of the SDK to reduce crashes and unexpected behavior. Improved error handling mechanisms, especially for network-related issues. Optimized memory usage to prevent memory leaks during long sessions.

Change log: Сhrome extension sample developed; new camera and microphone access method based on the promise; sip proxy added; other minor fixes; Try new features by downloading our latest update of WebRTC SIP SDK for web.

Please download our latest update of WebRTC SIP SDK. Change log: – outgoing call connection timeout reduced – custom headers for invite messages made possible – customizable user-agent SIP field added – DTMF implemented – possibility to perform multiple simultaneous calls added – video/audio recording implemented – application interface changes (done more consistent, obsolete methods removed, parameters changed, renamings) – possibility to select camera and microphone added – tag added to TO SIP header field – customizable register expiration timeout – some of the deprecation warnings resolved – hold redone to be more flexible with different servers – various fixes – new developer manual written