AlqaTech WebRTC SDK
With AlqaTech WebRTC SDK for Mobiles it becomes very easy to integrate WebRTC based VoIP Calling in Application. Our WebRTC SDK is based on SIP. AlqaTech WebRTC SDK is compatible with all major SIP Servers like Asterisk, Kamailio, FreeSwitch, OpenSIPs etc
Why SIP based WebRTC SDK?
WebRTC can not work standalone, It needs some singling to initiate WebRTC Session. During coordination initial call information is exchanged between Calling Party, Server and Callee party. Once singling os completed then Audio / Video streams are connected between parties using WebRTC peer-connection APIs.
Session Initiation Protocol (SIP) facilitates interactive user sessions comprising of voice, video, IMs, etc. The major advantage of using SIP for VoIP is its interactivity and ease of operation. SIP can be a boon for the communication service providers as it bridges the gap between the conventional telephone and IP communication services. SIP trunking allows you to take your business global by allowing you to manage and access your business anywhere with an Internet connection.
WebRTC SDK with SIP
AlqaTech WebRTC SDK enables users to make WebRTC based Media session through SIP Singling. SIP Signalling is widely used by telecom operators globally. SIP has the capability to provide Audio Video calling session
- Compatible with Standard SIP Servers like Asterisk, Kamailio, OpenSIPs, FreeSwitch etc
- App to App to Audio Video Calling
- App to PSTN/GSM Network Calling
- Call Billing Servers
- STUN/TURN Servers
- SIP Based Integration with Legacy Telephony networks
- WebRTC to SIP and SIP to WebRTC
- Android and iOS Native SDKs
WebRTC SDK Android
Please find more details about WebRTC SDK for Android on following link.
WebRTC SDK iOS
Please find more details about WebRTC SDK for iOS on following link