NewAn iPhone app, calls between companies and streaming into meetings

Calls, meetings and chat, rewired for the web.

Mentor is a cloud PBX, HD video meetings and team chat in one multi-tenant platform — built on SIP, WebRTC and open standards, with a softphone that runs right in your browser and on your iPhone.

Start free See it live

Built on

SIP
WebRTC
Opus
SRTP
ICE

The product

Every conversation, one mentor.

Calls, meetings, chat and the contact center share one identity, one presence model and one API — so a ring, a meeting link and a message are all part of the same conversation.

01Calls

A full softphone that lives in a tab.

Place and answer calls straight from the browser over SIP on WebSockets — no plug-ins, no desk phone required. Hold, transfer and keypad work the way people expect, with live call-quality scores on every call. The same extension rings on your iPhone, even when the app is closed.

HD voice with G.722 wideband audio, encrypted with DTLS-SRTP
Blind and attended transfer, hold, mute and DTMF (RFC 4733)
Browsers, SIP desk phones and the iPhone app ring on the same extension
Forwarding, do not disturb and a voicemail greeting of your own

02SBC & media

An edge that keeps media honest.

Every call enters through a Kamailio session border controller that authenticates, rate-limits and routes per tenant. rtpengine anchors the media and bridges browser DTLS-SRTP to plain RTP for the PBX, so NAT, encryption and codecs are handled before a call reaches your dial plan.

Registrar, digest auth, flood limits and scanner bans at the edge
DTLS-SRTP ↔ RTP bridging with ICE, rtcp-mux and BUNDLE
SIP over TLS with SRTP for desk phones and the iPhone app, which a push wakes for a call
Tenant-aware dispatch to Asterisk with automatic failover

03Meetings

Meetings that pick up where calls leave off.

HD video rooms run on a LiveKit SFU with simulcast and an active-speaker spotlight. Turn any phone call into a video meeting in one click, admit guests from a lobby, let people dial in from a regular phone, and bring a camera or OBS in as a stream.

Active-speaker layouts, screen share and recording
Lobby with admit and deny, plus host controls
Call-to-video escalation and SIP phone dial-in
Streaming in from an encoder over WHIP (RFC 9725)

04IVR builder

Design call flows the way you sketch them.

Wire menus, business hours, HTTP lookups and transfers together on a visual canvas. Every publish is a new version, so you can test, ship and roll back call routing without editing a dial plan.

Menus, gathers, schedules, conditions and timers
HTTP request nodes to route on CRM data
Versioned publishing with one-click restore

05Queues & ACD

Queues your agents actually like.

Route callers to the right agent with longest-idle, round-robin, ring-all or skills-based strategies. Position announcements, wrap-up time and a real-time wallboard keep agents, supervisors and callers in the loop.

Atomic agent reservation: never two calls offered to one agent
Agent states, wrap-up and automatic away after missed offers
SLA, speed of answer and abandon rate on the wallboard

06Team chat

Chat that knows who's on a call.

Channels, DMs and threads with presence that reflects real call and meeting state, so you know who is free before you ring. Mentions, reactions, read receipts and files arrive instantly over Centrifugo.

Presence derived from calls, meetings and devices
Threads, quoted replies, mentions, reactions and read receipts
Full-text search across every conversation

support-escalations14 members

AC

Ana Costa10:42

Caller on line 2 says pressing 2 loops back to the main menu.

ML

Marcus LeeOn a call10:43

@Ana the support queue was paused for the release — resuming now.

AC

Ana Costa10:44

Callers are flowing again. Wait time back under 20s.

42

Priya is typing…Seen by 3

Message #support-escalations

07API & webhooks

Everything in the app is an API call.

The REST API is described by an OpenAPI 3.1 spec, errors are RFC 9457 problem details, and organization API keys authenticate with a single header. Subscribe to call, queue, meeting and chat events with signed Standard Webhooks.

OpenAPI 3.1 reference with interactive docs
Scoped organization API keys via x-api-key
Signed, retried Standard Webhooks deliveries

create-meeting.tsTypeScriptcURL

1const res = await fetch(`${base}/api/v1/meetings`, {2  method: 'POST',3  headers: {4    'x-api-key': process.env.SIDEKICK_API_KEY,5    'content-type': 'application/json',6  },7  body: JSON.stringify({8    title: 'Escalation · order 4821',9    kind: 'instant',10  }),11})12const { joinUrl, dialIn } = await res.json()

call.endedStandard Webhooks signed

call.ended· Standard Webhooks 200 · signed

POST /hooks/sidekickwebhook-id: msg_2Lq8vT0x…webhook-timestamp: 1790843112webhook-signature: v1,K5oZfz…={ "type": "call.ended",  "data": { "direction": "inbound", … } }

What's new · October 2026

New on the platform.

The latest additions, all live today: an iPhone app that rings when it is closed, calls between companies, streaming into meetings and recording to a recorder of your own.

Sidekick Tel for iPhoneBeta · TestFlight
Your extension on your iPhone, over SIP-TLS with encrypted audio. The platform wakes the app with a push, so calls ring even when it is closed.
Calls between companiesBy address
Call anyone on Mentor by address: a handle such as maya@sidekick.tel, or an extension at their company. Each company chooses whether to take part.
Forwarding and do not disturb
Forward calls always, when busy, on no answer or when you cannot be reached, and switch on do not disturb. Voicemail plays a greeting of your own.
Stream into meetingsWHIP · RFC 9725
Bring OBS, a camera or a mixer into a live meeting. The host is given an address and a token, and the stream joins as a tile of its own.
Record to your own recorderSIPREC · RFC 7866
Keep recordings in Mentor, or have recorded calls sent to your company’s recording server as they happen — a choice each company makes.
Opus and G.729 at the edgeCodecs
The iPhone app talks Opus, with G.729 as an option. The media engine converts both to wideband G.722 at the edge, before a call reaches the PBX.
Session timersRFC 4028
Calls are refreshed while they last, and a call whose far end has vanished is cleared instead of lingering until someone notices.
Click-to-call
Pick a number in the app and Mentor rings your phones first, then places the call for you — on your own line, with your own calling rights.

Platform

Seven servers. One platform.

Battle-tested open-source engines do the real-time heavy lifting; typed TypeScript services carry the product logic around them. No proprietary SIP stack, no black boxes.

PBX
Asterisk 22 LTS · ARI
Call routing, extensions, ring groups, forwarding, feature codes, voicemail, recording and call records — driven over ARI by a typed router.
SBC
Kamailio
The SIP edge: WebSocket and UDP/TCP/TLS, registrar and auth, NAT handling, scanner bans, per-tenant routing, and the push that wakes a phone for a call.
Media
rtpengine
Anchors every edge stream, bridges DTLS-SRTP, SRTP and RTP, converts Opus and G.729 for mobile, and handles prompts, hold music and recordings.
Meeting
LiveKit SFU
Video rooms with lobby, host controls, recording, phone dial-in, streaming in over WHIP and one-click call escalation.
IVR
Flow engine on ARI
Runs versioned call-flow graphs: menus, DTMF, schedules, conditions, HTTP lookups and transfers.
Queue / ACD
Valkey-backed dispatcher
Hold music, position announcements, routing strategies, agent states, wrap-up and live statistics.
IM
Centrifugo
DMs, channels, threads, reactions and presence, plus the bridge that streams events to every browser.
Control plane
Fastify API · Next.js app
Tenants, roles, provisioning, the public REST API and webhooks — and the app you're looking at.

Underneath:Postgres · Valkey · NATS JetStream · S3-compatible storageContainerised with Docker Compose for Ubuntu

Standards

Built on open standards.

Mentor interoperates because it implements the specs, not a proprietary dialect. Here is what ships today.

SIP signaling

The same SIP your desk phones speak — over UDP, TCP, TLS and WebSockets from the browser.

SIPRFC 3261
Offer/answerRFC 3264
Symmetric response (rport)RFC 3581
Reliable provisional (PRACK)RFC 3262
UPDATERFC 3311
Reason headerRFC 3326
INFORFC 6086
REFER (transfer)RFC 3515
Session timersRFC 4028
SIP OutboundRFC 5626
SIP over WebSocketRFC 7118
WebSocketRFC 6455

SDP & RTP

Session description and real-time transport.

RFC 8866SDP
RFC 3550RTP / RTCP
RFC 3551A/V profile
RFC 4585RTCP feedback
RFC 5761RTP/RTCP mux
RFC 5506Reduced-size RTCP
RFC 9143BUNDLE

Media security

Browser, phone and app media is encrypted end to the edge.

RFC 3711SRTP
RFC 4568SDES keys
RFC 5763DTLS-SRTP framework
RFC 5764DTLS-SRTP
RFC 8827WebRTC security

NAT traversal

Connectivity that survives real networks.

RFC 8445ICE
RFC 8838Trickle ICE
RFC 8489STUN
RFC 8656TURN

WebRTC

Standards-track browser real-time media.

RFC 9429JSEP
RFC 8825–8835WebRTC suite
RFC 8853Simulcast

Codecs

Wideband voice, scalable video and out-of-band DTMF.

RFC 6716Opus
RFC 7587Opus over RTP
ITU-TG.711
ITU-TG.722 wideband
ITU-TG.729
RFC 4733DTMF events
VideoVP8 · H.264
VideoVP9 / AV1 SVC

Web & API

The product surface, held to the same bar.

W3CWCAG 2.2 AA
RFC 9457Problem details
OASOpenAPI 3.1
SpecStandard Webhooks

Desk phones

Lamps and message lights, the way phones expect them.

RFC 6665Event notification
RFC 3842Message waiting
RFC 4235Dialog events (BLF)

Recording & streaming

Calls out to your recorder, encoders into your meetings.

RFC 7866SIPREC
RFC 7865Recording metadata
RFC 9725WHIP

Sign-in

Single sign-on, passkeys and one-time codes.

SpecOpenID Connect
W3CWebAuthn passkeys
RFC 6238TOTP

On the roadmap: STIR/SHAKEN caller ID verification and attestation, SFrame end-to-end encryption for meetings (RFC 9605), T.38 fax and next-generation emergency calling (NG911).

Interaction budget<150 msINP target for every marketing page, and under 100 ms for chat and the dialpad.
Accessibility baselineWCAG 2.2 AABoth themes, full keyboard paths, and non-drag alternatives for every drag interaction.
Encrypted by defaultDTLS-SRTPBrowser calls and meetings carry encrypted media from the very first hop.
Self-hostableUbuntu + DockerEvery server ships as a container. Run Mentor in our cloud or on your own hardware.

Get started

Give your team a mentor.

Create a workspace, invite your team, claim extensions and place your first browser call in minutes. Free for 14 days — no credit card, no desk phones required.

Start free See pricing

Calls, meetings and chat, rewired for the web.

Every conversation, one mentor.

A full softphone that lives in a tab.

An edge that keeps media honest.

Meetings that pick up where calls leave off.

Design call flows the way you sketch them.

Queues your agents actually like.

Chat that knows who's on a call.

Everything in the app is an API call.

New on the platform.

Sidekick Tel for iPhoneBeta · TestFlight

Calls between companiesBy address

Forwarding and do not disturb

Stream into meetingsWHIP · RFC 9725

Record to your own recorderSIPREC · RFC 7866

Opus and G.729 at the edgeCodecs

Session timersRFC 4028

Click-to-call

Seven servers. One platform.

PBX

SBC

Media

Meeting

IVR

Queue / ACD

IM

Control plane

Built on open standards.

SIP signaling

SDP & RTP

Media security

NAT traversal

WebRTC

Codecs

Web & API

Desk phones

Recording & streaming

Sign-in

Engineering commitments

Give your team a mentor.

Calls, meetings and chat, rewired for the web.

Every conversation, one mentor.

A full softphone that lives in a tab.

An edge that keeps media honest.

Meetings that pick up where calls leave off.

Design call flows the way you sketch them.

Queues your agents actually like.

Chat that knows who's on a call.

Everything in the app is an API call.

New on the platform.

Sidekick Tel for iPhoneBeta · TestFlight

Calls between companiesBy address

Forwarding and do not disturb

Stream into meetingsWHIP · RFC 9725

Record to your own recorderSIPREC · RFC 7866

Opus and G.729 at the edgeCodecs

Session timersRFC 4028

Click-to-call

Seven servers. One platform.

PBX

SBC

Media

Meeting

IVR

Queue / ACD

IM

Control plane

Built on open standards.

SIP signaling

SDP & RTP

Media security

NAT traversal

WebRTC

Codecs

Web & API

Desk phones

Recording & streaming

Sign-in

Engineering commitments

Give your team a mentor.