Voice ProgramsVoice FeaturesVoice CoverageVoice APIVoice Use CaseVoice PricingClientsWebRTCGlobal Audio ConferenceSIP Interface
Video ProgramsFlexible SIP TrunkingNetwork Traversal ServiceSIP Registration
Voice ProgramsVoice FeaturesVoice CoverageVoice APIVoice Use CaseVoice PricingClientsWebRTCGlobal Audio ConferenceSIP Interface
Video ProgramsFlexible SIP TrunkingNetwork Traversal ServiceSIP Registration
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Steps in Building - WebRTC Gateway
Bridge conventional IPs with FONE API and improve communication in web-enabled device over any carrier network or SIP.
SIP Connect
Integrate WebRTC in your contact center and improve the user experience. Get acquainted with FONE API SIP Interface, FONE API Client, webhooks, and FONE API to route calls from your PBX to a SIP domain that you can easily set up. The application directs foneAPIML which WebRTC endpoint to communicate with.
STEP 1
Build A Sip Interface with FONE API
Route PBX calls to a SIP domain using FONE API interface. You can also enable connection over SRTP/TLS and set up your own safety and security guidelines.
STEP 2
Setup Your Call Center
Using FONE API, setup your VoIP infrastructure to direct incoming calls on a specific number to your new SIP domain. Your web servers will receive a callback via HTTP every time FONE API gets a call.
STEP 3
Link SIP Interface and WebRTC Clients
Configure your application web server to call numbers from the SIP-enabled domain. It will then communicate with foneAPIML to determine which client username or SIP username to direct the call.
STEP 4
Incorporate FONE API SDKs to Javascript Web APP
foneAPIML users sign up using a token which specifies the client or SIP username. FONE API SDKs include functions that are initiated once a call comes in.
Connect Over PSTN
Combining WebRTC into your existing contact center via PSTN entails becoming acquainted with FONE API’s Programmable Calls, foneAPIML, U clients, and webhooks. Incoming calls are obtained from your PBX system to the contact numbers you created on foneAPIML, and your application tells foneAPIML which WebRTC endpoint to direct the call.
STEP 1
Give each WebRTC Agent a Specific Number
Every WebRTC-enabled agent should have its unique phone number for FONE API to accurately pick up calls from your contact center.
STEP 2
Setup your Call Center
Send every call that your call center normally assigns to a WebRTC agent to your new FONE API number. The moment YC receives these calls, your web server receives a webhook.
STEP 3
Link SIP Interface and WebRTC Clients
Your app arranges the number with the WebRTC agent, and answers with foneAPIML to identify the username to call.
STEP 4
Incorporate FONE API SDKs to Javascript Web APP
FONE API users sign up using a token which specifies the client or SIP username. FONE API SDKs include functions that are initiated once a call comes in.
THE FONE API EDGE
Redundancy
Automated failover ensures that you have 99.95% uptime SLA without the need for a maintenance window.
Scalability
Use existing apps to new markets by configuring features for compliance and localization.
Multi-channel
Use a single platform for voice, SMS, video, authentication, chat, and more.
Without hassles
Gain free support, have the freedom to scale your business, and market faster with pay-as-you-go.
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