SIP.js Gains Support for In-Band DTMF (Beta) in Latest Version 0.10.0

Our engineering team just released a few important updates to SIP.js in version 0.10.0. The main highlights of this release: adding support for in-band DTMF and attaching media via Session Description Handler Observer.

Read More

Topics: developer, Business Technology, sip.js, WebRTC

SIP.js v0.8.0 Supports All Major Browsers and Renegotiation

We recently released version 0.8.0 of SIP.js, which allows codec renegotiation to occur during WebRTC calls. This enables several new features, including music on hold and the ability to add video to an ongoing audio call. With this update, SIP.js is also now supported by all major browsers.

Read More

Topics: Business Technology, sip.js, WebRTC

Build and Manage WebRTC Applications with SIP.js and Callstats.io

These days, developers working with real-time web-based communications have more resources than ever before. But what tools are best for analyzing the performance of WebRTC apps? We recently integrated with callstats.io, a cloud-based monitoring and management service for WebRTC so that developers using SIP.js can monitor and analyze key metrics and errors on their calls.

Read More

Topics: Business Technology, sip.js, WebRTC

Best CPaaS Providers: 5 Choices for Voice, Video, and Messaging

Text message reminders of doctor appointments, click-to-dial phone calls with support agents, and in-app streaming videos are just a handful of examples that highlight the pervasiveness of real-time communications (RTC) in apps that span many different categories.

Consumers expect easy access to voice, video, SMS, and other RTC features in a wide variety of applications. Even apps that aren't communications-based are expected to offer RTC as a way to enhance the user experience.

Read More

Topics: developer, sip.js, WebRTC, video call

SIP.js Update: Video Conferencing & Secure Calling Added

Our engineering team recently enabled the latest version of FreeSWITCH 1.6.14 for SIP.js. This update will help developers build real-time communications apps with expanded video and security capabilities.

Read More

Topics: Business Technology, sip.js, WebRTC, freeswitch

Detecting Abnormal VoIP Traffic, WebRTC, and SIP at ClueCon

Last week, three members of the OnSIP engineering team attended ClueCon 2016 in Chicago for three days of coder games, round table panels, and informative sessions on real-time communications. As proponents of open source technology, we've been attending ClueCon for many years.

Read More

Topics: SMB Leadership, WebRTC, sip.js, SIP

OnSIP Speakers Talk SIP and WebRTC at Real Time Web Solutions Conference

Our CTO John Riordan and Lead Engineer James Criscuolo both spoke on panels at the 2016 Real Time Web Solutions Conference on August 3rd. OnSIP was a silver sponsor of the event, which took place August 1-4 at NYU's Kimmel Center in New York City.

Read More

Topics: SMB Leadership, WebRTC, sip.js, SIP