VoIP and Video Servers on a VPS: Voice, Video Calls and Streaming

Voice and video punish a slow network in a way most server workloads don’t. A web page that loads a little late still loads. A call with late or uneven packets breaks up, and a live stream that can’t keep up drops frames. This section covers running those services yourself on a VPS, and sizing the server and network to suit them.

What will you find in this section?

Practical guides for self-hosted voice, video calling and live streaming. On the voice side, that means PBX setups with Asterisk and FreePBX, plus voice chat servers like Mumble and TeamSpeak. For video calls, it means Jitsi Meet, WebRTC, and running your own TURN and STUN servers so calls connect through NAT. For streaming, it means SRT and RTMP relays and restreaming one feed to several platforms at once.

Expect setup steps on Debian or Ubuntu, firewall and port notes, and plain sizing advice for each workload. We’ll add guides here over time.

Why do latency and jitter matter so much?

Real-time audio has a tight delay budget. ITU-T G.114 recommends keeping one-way delay under 150 ms for most voice applications. Jitter, the variation in that delay, makes things worse, because the receiving end has to buffer to smooth it out.

Where the server sits sets the baseline. Our servers run in the IDACORE Boise datacenter in Boise, Idaho: under 5 ms to Treasure Valley businesses, 14 ms to Salt Lake City, 22 ms to Portland and 23 ms to Seattle.

The standard port is 1 Gig. A 2.5 Gig port is a $5/mo add-on, and 10 Gig is a $15/mo add-on. Transfer is an allowance: if you go over it, the port is throttled, never billed, and extra transfer is $1.50/TB.

For a PBX, a voice chat server or a stream relay, the NVMe VPS for voice and video starts at $10/mo, priced at $5/GB RAM. Dial it in to the size your workload needs.

Part of the ADTAQ blog.

Topics in VoIP and Video

  • VoIP

    Running a PBX or voice chat server on a VPS with Asterisk, FreePBX, Mumble or TeamSpeak.

    No guides yet

  • Video Calls

    Self-hosted video calling with Jitsi Meet, WebRTC and your own TURN and STUN servers.

    No guides yet

  • Live Streaming

    SRT and RTMP relay servers and restreaming your live video to several platforms.

    No guides yet

Latest in VoIP and Video

No guides here yet. The first ones are being written now.

Frequently asked questions.

Yes, a VPS can run a VoIP server such as Asterisk or FreePBX for phone calls, or Mumble or TeamSpeak for voice chat. You'll need to open the right ports in your firewall and keep the system patched. Pick a server location close to the people who'll use it, because network distance adds delay to every call.

Put a voice or video server as close to your users as you can, because network distance adds delay to every packet. ADTAQ's servers are in Boise, Idaho, with measured latency of under 5 ms to Treasure Valley businesses, 14 ms to Salt Lake City, 22 ms to Portland and 23 ms to Seattle. If most of your users are in the Northwest, that keeps the network leg of each call short.

Many self-hosted video setups need a TURN server, because WebRTC can't always connect participants directly. Strict NATs and firewalls can block the direct path, and a TURN server relays the media when that happens so the call still connects. A STUN server only helps clients discover their public address. Jitsi Meet and other WebRTC tools work with both, and running your own keeps call traffic on a server you control.