You can use your FreeCall account with a wide range of SIP-compatible devices and applications. Simply sign up for an FreeCall account and use your FreeCall username and password to register your device with our SIP servers.

Network Setup

SIP configuration

Supported SIP devices

Softphones for Windows, macOS and Linux (such as Linphone, Zoiper, MicroSIP and Bria)

Mobile SIP apps for Android and iOS (such as Linphone, Zoiper and Groundwire)

IP desk phones (such as Yealink, Grandstream, Snom and Fanvil)

SIP ATAs (Analogue Telephone Adapters) (such as Grandstream HT801/HT802 and Cisco SPA112)

IP PBXs (such as Asterisk, FreePBX, 3CX and FreeSWITCH)

SIP routers and VoIP gateways (such as MikroTik, DrayTek and Cisco)

Supported SIP protocols

SIP over UDP (default)

SIP over TLS for encrypted SIP signalling

Note: SIP over TLS encrypts the SIP signalling between your device and our servers. Voice media is transmitted using standard RTP. SIP over TLS uses the standard port 5061, which is automatically selected by most SIP clients when TLS is enabled.

Supported audio codecs

G.711 U-law (PCMU) – High quality (64 kbps)

G.711 A-law (PCMA) – High quality (64 kbps)

G.722 – HD Voice (wideband audio)

G.729a – Low bandwidth (8 kbps)

G.726 – Medium bandwidth (32 kbps)

iLBC – Optimized for unreliable network connections

Note: During call setup, your SIP device and our servers automatically negotiate the best available codec supported by both sides. In most cases, no manual codec configuration is required.

Troubleshooting audio issues

If you experience one-way audio or other audio-related problems, try the following:

Configure a STUN server (if supported by your SIP device or application): Server: stun.freecall.com Port: 3478

Prefer the G.711 codec for the best compatibility and call quality.