Linphone is a free, open-source SIP softphone that makes HD voice and video calls, sends instant messages, and hosts conferences over any standard VoIP account, including a free SIP address. It suits users and businesses that want encrypted calls without a proprietary cloud.
Linphone is an open-source softphone that places SIP calls, video calls, and instant messages over standard VoIP accounts instead of a proprietary service. You register one or more SIP accounts in the app, then dial usernames, SIP addresses, or traditional numbers through your provider's gateway. Calls run over Opus and modern video codecs, with echo cancellation and stream adaptation that keeps audio usable on weak connections, and an on-screen quality indicator shows what the network is doing. The same window holds the contact list, chat history, and call log, so day-to-day communication stays in one place. Because it speaks the same protocols as IP-PBX systems, hardware SIP phones, and hosted providers, Linphone fits an existing phone setup rather than replacing it.
Most SIP softphones assume you already have a service, while Linphone includes a free SIP address from the project itself, so two people can test a call within minutes. Scale changes the setup rather than the app. One-to-one messaging and simple audio conferences work on virtually any standard server, while group chat, file sharing, read receipts, and video meetings with dozens of participants need the developer's Flexisip server behind them. Deployment for teams leans on remote provisioning: an administrator hands out a QR code or link that fills in account details and preferences. Users comparing voice tools often come from gaming-focused apps such as TeamSpeak 3 Client, where the emphasis is low-latency group voice rather than telephony, phone numbers, and interoperability.
The biggest advantage of Linphone is that it refuses to lock you into a vendor. Because it speaks standard SIP, the same app works with a free public account, a self-hosted PBX, or a business telephony provider, and switching providers means editing a field rather than changing software. Security is unusually deep for a free client: calls can be encrypted with SRTP, ZRTP, or DTLS-SRTP, messages with the project's own end-to-end encryption, and post-quantum key exchange is already included so traffic captured today stays unreadable later. Call handling covers the practical ground teams need, including blind and attended transfers, hold with music, multi-call management, recording with in-app playback, auto-answer for support desks, and a quality indicator that turns vague complaints into measurable numbers. Video reaches 1080p with modern codecs, streams adapt when the network dips, and calls survive a switch from Wi-Fi to mobile data. Administrators also save time through QR-code or link-based provisioning and LDAP or CardDAV contact sync.
Key capabilities and highlights.
Calls use Opus audio and modern video codecs such as H.264, H.265, AV1, and VP8, with software or hardware echo cancellation and forward error correction to smooth packet loss. A quality indicator and call statistics expose bandwidth, jitter, and the codec in use, so problems get diagnosed instead of guessed at.
Chat covers one-to-one and group conversations with delivery and read receipts, replies, forwarding, reactions, voice notes, and file or document sharing, plus a media space for finding earlier attachments. Ephemeral messages delete after being read or after a set period, keeping sensitive details out of long-lived history.
Audio and video conferences run through the developer's SFU server for smooth meetings with many participants, while callers on third-party SIP servers can still merge several calls into a simple audio conference. Screen sharing and audio-only, active-speaker, or mosaic layouts keep larger meetings readable on any screen.
Media can be encrypted with SRTP, ZRTP, or DTLS-SRTP, while messaging uses its own end-to-end scheme, and encryption can be made mandatory so calls never fall back to plain media. Trust verification adds a visible badge once a contact's device is confirmed.
Blind and attended transfers, holding with music, multi-call handling, recording with in-app playback, auto-answer, DTMF tones, and message waiting indicators cover the daily routines of a help desk. Bluetooth headset support and a searchable call history make it practical for staff who spend hours on the phone.
The address book can be filled manually, searched against an LDAP directory, or synchronized through CardDAV, so corporate contacts appear without typing them in. Presence updates show whether colleagues are reachable before you dial, and multi-account support keeps work and personal lines apart.
Administrators can generate a QR code or a one-click link containing account credentials and app preferences, which users import to configure everything at once. Individuals can register several SIP accounts using different transports, then pick which line an outgoing call should use.
The client follows SIP and related IETF standards, including ICE, STUN, and TURN for NAT traversal, which lets it register with most IP-PBX systems, hosted providers, and hardware SIP phones. That interoperability means an existing number and phone infrastructure keep working while the app on top changes.
Browse historical releases and previous builds.
Common questions about Linphone.
Linphone is a SIP softphone, so people use it to make and receive voice and video calls, exchange messages, and hold conferences through a VoIP account. It registers with a hosted provider, a company PBX, or the project's free SIP service, which makes it a workable replacement for a desk phone on a computer, tablet, or phone.
Yes. The application is open source and free for everyday use, and it needs no subscription to place calls as long as you have a SIP account to register. Organizations that want branded builds, custom development, or guaranteed support buy those services separately from the app itself.
You need a SIP account, but not necessarily a paid provider. The project offers a free SIP address, and many VoIP services sell inexpensive accounts that include a real phone number. If your workplace runs an IP-PBX, IT supplies the login details and Linphone registers with that server directly.
Yes. Calls can be encrypted with SRTP, ZRTP, or DTLS-SRTP depending on what both ends negotiate, and messaging has its own end-to-end encryption layer. Trust verification shows a badge once a contact's device is confirmed, and settings can require encryption so a call never falls back to unprotected media.
Meetings hosted on the developer's SFU conferencing server can scale to roughly 50 participants with screen sharing and several layout modes. On third-party SIP servers, the app still supports simple audio conferencing by merging several calls, which suits small team huddles better than a large webinar.
It follows SIP and the related standards around media and NAT traversal, so it registers with most IP-PBX systems, hosted providers, and hardware SIP phones. Final compatibility depends on which extensions a specific server enables, but the client's interoperability focus keeps migration surprises to a minimum.
Basic one-to-one messaging uses the standard SIP SIMPLE protocol, so it works with almost any SIP account. Group conversations, file sharing, read receipts, reactions, and ephemeral messages depend on the developer's Flexisip server, so confirm your server setup before promising those capabilities to a team.
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