This image provides a fully installed FreeSWITCH server on an Azure-optimized Linux distribution with cloud-init and Azure Linux Agent preconfigured. It is built for reliability, security, and low-latency media handling so you can stand up SIP, RTP, and WebRTC services quickly in your Azure subscription.

Who is it for?

Telecom/VoIP engineers and MSPs shipping SIP voice solutions on Azure
Contact center and UC providers needing media, IVR, and conferencing
Developers prototyping RTC features with SIP/WebRTC backends

Standing up FreeSWITCH from scratch is time-consuming and error-prone. This image removes guesswork with a hardened baseline, Azure networking guidance, and ready-to-use defaults so teams can move from provisioning to call flow testing quickly.

Key value (benefits over features)

Faster time to dial tone: Preinstalled FreeSWITCH with sensible defaults, so you can register endpoints and place test calls quickly.
Production-ready baseline: Firewall/NSG guidance, SIP/RTP ranges, log rotation, automatic security updates, and systemd services configured.
Azure-tuned: Cloud-init, Azure Linux Agent (waagent), Accelerated Networking support on compatible NICs, VM size recommendations, and time sync.
Secure by default: SSH key login, OS hardening, optional TLS for SIP, and guidance for locking down RTP ranges and management access.
Operate with confidence: Optional 24x7 support plans, runbooks, and upgrade guidance to keep you current with upstream security patches.

What’s included

FreeSWITCH (1.10.x series) installed and enabled as a system service
Common modules for SIP, media, IVR, conferencing (custom modules can be enabled later)
Log rotation, journald persistence, and basic health checks
Cloud-init for first-boot configuration and idempotent setup
Azure Linux Agent for VM lifecycle and diagnostics

Recommended VM sizes

Development & small PBX: B2s, B2ms
SMB production or IVR: D2s v5, D4s v5
High-concurrency media: D8s v5 or higher, consider Accelerated Networking on supported NICs. Use Premium SSD for consistent I/O on busy workloads. Place RTP-heavy services close to your SIP carriers or SBCs to reduce jitter.

Networking & ports (configure in your NSG)

SIP Signaling: UDP/TCP 5060 (and 5061 for TLS if enabled)
RTP Media: UDP 16384–32768 (adjust to your policy)
SSH Administration: TCP 22 (limit to your IP or jump host)
Optional WebRTC/WS: TCP 7443/5066 if you enable those modules

Quick start (5 steps)

Deploy the image to your resource group and choose a recommended VM size.
Assign a static public IP (if required) and attach to a dedicated subnet with an NSG following the port guidance above.
SSH to the VM using your key and verify service status: systemctl status freeswitch.
Set SIP domains and dial plan in your FreeSWITCH configs; reload with fs_cli -x "reloadxml".
Open NSG ports for SIP/RTP testing and register a softphone to place a test call.

Common use cases

PBX and enterprise telephony
SBC/media relay in front of carrier trunks
IVR, DTMF bots, and call routing
Audio conferencing and call recording
WebRTC prototypes and RTC microservices

FreeSWITCH on Azure - Production-Ready SIP ServerSolve DevOps

FreeSWITCH on Azure - Production-Ready SIP Server

Solve DevOps

FreeSWITCH on Azure - Production-Ready SIP Server

Solve DevOps

FreeSWITCH on Azure - Production-Ready SIP Server
Solve DevOps