Bring Every Participant In: Telephony in Digital Samba

Today’s virtual meetings go beyond video and chat. They are about inclusivity, accessibility and seamless communication across distance. Digital Samba’s telephony capabilities make this possible by allowing participants without internet access or apps to join video conferencing sessions simply by dialling in.

Table of contents

1. Telephony — a simple way to join meetings
2. Under the hood: telephony with SIP/PSTN integration
3. Practical tips for success
4. Why telephony still matters
5. Getting started

Telephony — a simple way to join meetings

At its core, telephony in Digital Samba allows participants to join meetings by phone. Instead of clicking a link, an attendee can dial a standard phone number and enter a PIN to connect to the audio portion of the session. Once connected, they appear in the participant list like any other attendee, complete with a clear phone icon and caller information where available. Moderators can mute or unmute phone participants directly within the session, and callers can raise their hand to request to speak without disrupting the flow of the meeting.

However, a couple of limitations apply when telephony is enabled:

• Breakout rooms aren't available during the call. This ensures call audio remains stable and prevents routing issues between multiple sub-sessions.
• Only a single telephony bridge can be connected to a room at once. This simplifies call management and avoids conflicts between different phone providers.
• End-to-end encryption (E2EE) isn't supported in sessions where telephony is active. This is because traditional phone networks cannot support encrypted media in the same way as web connections.

To enable this feature, telephony must first be included in your plan (see our pricing plans for details). Once available, it can be enabled directly in the room settings. Hosts configure a dial-in phone number together with a PIN (DTMF) sequence to control access, providing a simple but effective security layer.

The DTMF sequence supports timing controls, allowing pauses for IVR prompts so the system can reliably navigate automated phone menus.

Under the hood: telephony with SIP/PSTN integration

For organisations that require a more custom and robust telephony solution, including integration with existing business telephony systems, Digital Samba provides a detailed SIP/PSTN Telephony integration guide. The guide explains how to build a bridge between traditional phone networks and Digital Samba rooms using your preferred voice API provider, such as Twilio, Sinch, Vonage or Plivo.

At a high level, the integration works like this:

• Voice API provider: Handles incoming phone calls, IVR prompts and conferencing with the public switched telephone network (PSTN). This is the entry point for callers joining from landlines or mobile phones, and where PIN validation typically occurs.
• SIP trunk: Connects your voice provider to Digital Samba. It acts as the secure communication channel for real-time audio exchange, requiring IP whitelisting and authentication configured with Digital Samba support.
• Middleware application: Your app that routes calls, manages PIN logic and mediates between the voice API and Digital Samba. This layer gives you full control over call flows, conference creation, user mapping and synchronisation between both platforms.
• Digital Samba room: The virtual space where your telephony and web participants meet. Phone users appear alongside browser participants for a unified meeting experience, with full moderator controls available.

How the connection flow works

When a moderator with the "Manage phone users" permission clicks to connect telephony, Digital Samba's SIP server initiates a call to your configured phone number. The system then sends the pre-configured DTMF sequence (the PIN) to navigate the IVR and join the correct conference. Once connected, room audio flows bidirectionally, allowing web participants to hear phone callers, and phone callers to hear the room.

Your middleware application plays a crucial role here. It maintains the mapping between Digital Samba rooms, voice API conferences and participant PINs. When phone callers dial in and enter their PIN, your middleware validates them, routes them to the correct conference, and then notifies Digital Samba via the Phone User API. This notification should only happen after the SIP bridge is connected,  thus ensuring phone users can actually hear room audio before they appear in the participant list.

Behind the scenes, your middleware keeps track of which phone callers are in which conference, and it updates Digital Samba using the Phone User API. This ensures moderators can see who has joined, mute participants, manage raised hands and control the call just like they would for any web-based user.

Practical tips for success

If you're planning to integrate telephony with Digital Samba:

• Contact Digital Samba support early to configure SIP trunking and authentication. Early coordination helps avoid delays and ensures your SIP trunk IP addresses are whitelisted before development begins.
• Build your middleware to manage calls efficiently, including IVR prompts and PIN validation. A well-designed call flow significantly improves the caller's first experience and reduces support queries.
• Use webhooks from Digital Samba to keep your system in sync with room events. This allows your application to react instantly to joins, leaves, mute actions and session lifecycle events.
• Plan your PIN and conference management strategy carefully. Voice API providers don't inherently know about conferences until you create them — your middleware needs to generate PINs, map them to rooms and handle routing logic.
• Connect telephony early in sessions when phone participants are expected. Phone callers who dial in before the SIP bridge is connected will be waiting in your voice conference without hearing the room audio.

Why telephony still matters

Even with powerful video and chat tools, there will always be situations where a participant:

• Has unreliable internet connectivity. Dial-in access ensures they can still participate without dropped audio or lag, using stable phone networks instead.
• Is joining from a location where video apps aren't accessible. This is common in secure corporate environments, hospitals or regions with restricted networks.
• Prefers to dial in by phone for convenience or familiarity. Some users simply find phone calls faster and more comfortable, especially when joining while travelling.
• Needs to join from a device without a camera or microphone. Phone dial-in provides a reliable fallback when hardware or software issues prevent web access.

Getting started

By offering both simple per-room telephony enablement and a fully customised SIP integration option, Digital Samba gives teams the flexibility to include every participant, whether they're joining from a browser, mobile device or a traditional phone line.

Ready to bring telephony to your Digital Samba integration? Start by reviewing our Telephony integration guide for the complete API reference, or contact our support team to begin configuring your SIP trunk.