Take control over your app call flow including call forwarding and routing, collecting input, playing music, recording and storing calls,, leaving a voicemail, sending an email and more.
RestcommONE provides robust audio and video conferencing bridge functionality for your collaboration needs. You can create room-based and ad-hoc conferences with personal identification (PIN) codes that offer full conference room controls for a moderator including muting and dropping participants.
RestcommONE provides audio signaling for all layers of the voice and video protocol stack using SIP Servlets and JSLEE via RESTful APIs. Included resource adaptors for SIP, SS7, SMPP, Diameter and other telecom protocols enable companies to connect to the mobile network worldwide
RestcommONE includes built-in and programmatic media recording and playback capabilities. Using Visual Designer or APIs, your team can implement features for recording voicemail and meetings, storing and forwarding messages and even transcription for voice to text.
Restcomm enables apps to implement intelligent Automatic Call Distribution (ACD) based on real-time information about the background of the callers. Features such as language, skills, time of the day and location can be easily implemented.
Restcomm leads the open source projects implementing three key Java SIP standards including JAIN SIP (JSR 32) and SIP Servlets (JSR 289, JSR 359), as well as JAIN SLEE (JSR 240) with SIP Resource Adaptor.
Create synthesized audio messages that are delivered in a language, accent and gender configured for your users.
Mobile, fixed, and toll-free international phone numbers worldwide.
Automatic speech recognition integration with services like Google Speech and IBM Watson enable you to create applications and services that listen to users in more than 150 languages. Imagine users driving your next app or service by talking to it!