/// TELECOM · VOIP & WEBRTCVoIP & WebRTC App Development
Voice and video calling built into your apps and browsers — SIP and WebRTC, calls to ordinary phone numbers, routing and recording — engineered for the networks and phones people actually use.
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TRUSTED BY TEAMS
A VoIP demo works on office Wi-Fi. A VoIP product has to work on a patchy mobile connection, behind a corporate firewall, on a locked phone with the app closed, and when the user walks from Wi-Fi onto 4G mid-call.
Sodio provides VoIP software development and WebRTC development for companies building calling into their products and for providers building voice services — voice and video in apps and browsers, calls to the phone network, routing, recording and the compliance that comes with them.
/// IS CUSTOM RIGHT FOR YOU?
When a Ready-Made Phone System Is Enough
If what you need is a business phone system for your own staff, buy one — established cloud phone systems handle it well. A VoIP development company earns its place when calling is part of the product you build.
A ready-made phone system fits when
- Your staff need desk phones and a mobile app
- Standard call features are enough
- Calls go to and from ordinary phone numbers
- You are buying a service
Custom VoIP fits when
- Your customers need to call from inside your product
- Calling has to fit your own workflow and data
- Calls connect users, agents, devices and phone numbers together
- You are building one — as a UCaaS, CPaaS or ISP voice provider
/// WHAT IT COVERS
What VoIP and WebRTC Development Covers
VoIP software development covers far more than connecting two people. These are the parts a production calling product needs.
Voice and video calling
Group calls and conferencing
Phone network calling
Native mobile call experience
Browser calling
Call routing and IVR
Recording and transcription
Messaging and presence
Call quality monitoring
Security and compliance
/// WHAT HAPPENS IN A CALL
What Happens Between Tapping Call and Hanging Up
A user taps call
The app requests the call and checks the user is allowed to make it
The other side is alerted
Signalling over SIP or WebRTC rings the recipient’s device, even if the app is closed
A network path is found
NAT traversal finds a route between the two devices, relaying through a server when needed
Audio and video flow
Encrypted media sent with a codec that adapts to the network
The call reaches its destination
Delivered to an app, a browser, a queue or an ordinary phone number
The call finishes
Call record, recording and quality measures saved
/// WHY CALLING IS HARD
Why Calling Is Harder Than It Looks
Works with Twilio, Vonage and Agora, SIP trunking providers, and open-source media servers such as Asterisk, FreeSWITCH and LiveKit.
///AI IN CALLING
Where AI Helps Calling
AI is changing what a call can do after it ends — and, increasingly, while it is happening.
Transcripts and summaries
Every call transcribed and summarised, with action points, for users who need a record of what was agreed.
Noise suppression
Background noise removed in real time, so a call from a busy street sounds like one from a quiet room.
Live support for agents
For contact centres, answers and next steps suggested while the customer is still speaking.
Every call transcribed and summarised, with action points, for users who need a record of what was agreed.
Background noise removed in real time, so a call from a busy street sounds like one from a quiet room.
For contact centres, answers and next steps suggested while the customer is still speaking.
/// COMPARE
Communications API, Open-Source Stack or Custom VoIP Solutions
Most products should start on an API. We build on all three, and help you move between them when the numbers change.
/// COST AND TIMELINE
What Drives the Cost and Timeline
A working calling feature on real devices typically arrives within 30 days, with a production build in 90.
The cost moves with six things: how many platforms it runs on — iOS, Android, web and desktop; whether calls reach the phone network; the size of group calls; recording and transcription; the compliance involved, from emergency calling to recording consent and data location; and how many calls must run at once.
/// /// FAQ
Frequently Asked Questions
Decide first where calls need to go — only between your own users, or to ordinary phone numbers as well — and which devices they must work on. Then choose whether to build on a communications API or run your own calling infrastructure, build the calling experience on each platform, handle the difficult parts such as incoming calls on locked phones and calls through firewalls, and test on real networks rather than office Wi-Fi.
WebRTC is the open standard that lets browsers and apps make real-time voice and video calls without plugins. You need WebRTC development whenever calls happen inside a web page or app, rather than on a traditional phone system. It is often combined with SIP, the standard telephone networks use, so app users can reach ordinary phone numbers.
Most products should start on a communications API such as Twilio, Vonage or Agora: it is the fastest way to launch. Running your own stack starts to make sense when per-minute fees grow large, when data has to stay in a particular country, or when you need control the API does not give you. We will say which applies to you.
Because phones close apps running in the background to save battery. An incoming call has to wake the phone through a special push notification and hand the call to the phone’s own calling screen — CallKit on iOS and ConnectionService on Android. Getting this right is one of the most common gaps in VoIP apps, and one of the first things we build.
It depends on what the service does and where. Services that connect to the phone network usually carry emergency calling obligations — in the US including rules on sending a caller’s location with 911 calls, and in the UK access to 999 and 112. Recording calls requires consent, and the rules differ between US states and between countries. In the US, outbound calls to the phone network must carry verified caller ID under STIR/SHAKEN.
On every call, not by sampling. Each call records its audio quality score, jitter, packet loss and delay, so problems can be traced to a network, a device or a region, and fixed before users give up on the app.
/// RELATED SERVICES
Related Industries and Services
Routes where scope overlaps with VoIP and WebRTC.
/// GET STARTED
Start With the Call Your Users Need to Make
Tell us who needs to call whom, on which devices, and whether calls must reach ordinary phone numbers. We will prepare a free solution architecture — including whether to build on an API or run your own stack.