Telephony Providers
Llamenos supports multiple telephony providers through its TelephonyAdapter interface. You can switch providers at any time from admin settings without changing application code.
Supported providers
| Provider | Type | Pricing Model | WebRTC Support | Setup Difficulty | Best For |
|---|---|---|---|---|---|
| Twilio | Cloud | Per-minute | Yes | Easy | Getting started quickly |
| SignalWire | Cloud | Per-minute (cheaper) | Yes | Easy | Cost-conscious organizations |
| Vonage | Cloud | Per-minute | Yes | Medium | International coverage |
| Plivo | Cloud | Per-minute | Yes | Medium | Budget cloud option |
| Telnyx | Cloud | Per-minute | Yes | Medium | Developer-friendly |
| Bandwidth | Cloud | Per-minute | Yes | Medium | US carrier-grade |
| Asterisk | Self-hosted | SIP trunk cost only | Yes (via sip-bridge) | Hard | Maximum privacy |
| FreeSWITCH | Self-hosted | SIP trunk cost only | Yes (via sip-bridge) | Hard | High-volume |
Pricing comparison
Approximate per-minute costs for US voice calls (vary by region and volume):
| Provider | Inbound | Outbound | Phone Number | Free Tier |
|---|---|---|---|---|
| Twilio | $0.0085 | $0.014 | $1.15/month | Trial credit |
| SignalWire | $0.005 | $0.009 | $1.00/month | Trial credit |
| Vonage | $0.0049 | $0.0139 | $1.00/month | Free credit |
| Plivo | $0.0055 | $0.010 | $0.80/month | Trial credit |
| Telnyx | $0.005 | $0.009 | $1.00/month | Trial credit |
| Asterisk | SIP trunk rate | SIP trunk rate | From SIP provider | N/A |
Feature support matrix
| Feature | Twilio | SignalWire | Vonage | Plivo | Asterisk |
|---|---|---|---|---|---|
| Call recording | Yes | Yes | Yes | Yes | Yes |
| Live transcription | Yes | Yes | Yes | Yes | Yes (via bridge) |
| Voice CAPTCHA | Yes | Yes | Yes | Yes | Yes |
| Voicemail | Yes | Yes | Yes | Yes | Yes |
| WebRTC browser calling | Yes | Yes | Yes | Yes | Yes (SIP.js) |
| Webhook validation | Yes | Yes | Yes | Yes | Custom (HMAC) |
| Parallel ringing | Yes | Yes | Yes | Yes | Yes |
SIP bridge
Self-hosted providers (Asterisk, FreeSWITCH, Kamailio) are accessed via the sip-bridge service. Set the PBX_TYPE environment variable to select the backend:
PBX_TYPE=asterisk # Asterisk ARI
PBX_TYPE=freeswitch # FreeSWITCH ESL
PBX_TYPE=kamailio # KamailioHow to configure
- Navigate to Settings in the admin sidebar
- Open the Telephony Provider section
- Select your provider from the dropdown
- Enter the required credentials
- Set your hotline phone number in E.164 format (e.g.,
+15551234567) - Click Save
- Configure webhooks in your provider’s console
See individual setup guides: