SIP Server as a Service

SIP Server as a Service offers seamless access to cloud-based SIP server functionalities, enabling you to create modern VoIP communication between SIP devices and applications effortlessly.

With SIP Server as a Service, there’s no need to set up, manage, or maintain your own SIP infrastructure. Everything runs as a service in the cloud. The SIP Server as a Service includes SIP registrars, user databases, SIP proxies, media relays, and NAT traversal mechanisms. It also handles the usual challenges of SIP device connectivity, reliability, and security, so you don’t have to.

Unlimited scale. Carrier grade performance.​

iotcomms.io’s SIP Server is built for unlimited scalability, capable of handling any deployment size, user base, and geographic reach. It’s designed to support mission-critical applications and manage carrier-grade volume and concurrent calls.

The SIP Server as a Service features a pay-as-you-go model, offering usage-based pricing. As a SaaS solution, it eliminates development costs and hidden hardware or operational expenses.

Key Benefits

SIP Server as a Service offers several user benefits, including:

Built in NAT & Firewall Traversal

Out-of-the-box Security

Unlimited & Global Scale

Geographic Redundancy

Instant
WebRTC Access

Automatic Codecs Transcoding

Key Functions

SIP Server as a Service includes a number of functions, such as:

Serving as a multi-tenant RFC 3261 SIP Registrar, it accepts customer SIP device registrations. SIP user accounts and registration states are maintained in a location service.

Acting as a multi-tenant RFC 3261 SIP Proxy, it relays SIP signaling between different SIP user accounts registered within the customer SIP domain.

SIP devices can connect to the traditional telephony world to place and receive calls, thanks to the built-in bring-your-own SIP trunk functionality.

Includes a media relay and SIP logic to ensure seamless delivery of media flows end-to-end, even when SIP devices are behind NAT.

The SIP Server applies signaling and media encryption and, if necessary, hosted NAT traversal, ensuring constant connectivity and readiness of your SIP devices.

Notifications can be sent through the AWS SNS service or via HTTPS callbacks to your preferred service.

The SIP Server can act as a hosted Session Border Controller (SBC) by letting an internal service connect with SIP devices located on the public internet. It supports two modes:

  1. The service hosts registrations of devices and the internal system is connected via a SIP trunk.

  2. Proxy integration where the service relay requests originating from clients on the internet to the internal system.

The hosted SBC functionality allows changing the transport protocols for SIP signaling and media. This enables devices using SIP/Websocket for signaling and RTP/DTLS for media to connect with an internal system, even if the internal system uses SIP/TCP and RTP/UDP.

Various transport protocols can be mixed and matched:

  • SIP/TCP
  • SIP/UDP
  • SIP/TLS
  • SIP/Websocket
  • RTP/UDP
  • SRTP/UDP
  • RTP/DTLS

All SIP requests (REGISTER, INVITE, etc.) are authenticated using SIP Digest authentication. No user passwords are stored in clear text, only the HA1 is stored.

Alternatively, SIP requests can utilize a JSON Web Token provided in a header for authentication.

SIP signaling is encrypted using TLSv1.2 or later. SIP devices must support RFC 5626 and maintain the TLS session to allow inbound SIP signaling.

When media anchoring is required (e.g. to apply Hosted NAT traversal) the media streams are relayed without alteration, ensuring end-to-end encryption is maintained.

SIP devices must support SRTP (RFC 3711) together with an appropriate key negotiation method to support end-to-end RTP encryption (SDES RFC 4568 or DTLS-SRTP RFC 5764).

Supporting SIP over Websocket (WebRTC), it facilitates the interoperability of audio and video sessions between WebRTC and native SIP clients.

Discover the WebRTC Gateway functionalities

Automatic transcoding from one codec to another is provided, which proves valuable in scenarios where your SIP devices support codec X while your trunk provider supports only codec Y.

Supported codecs:

  • PCMA
  • PCMU
  • OPUS
  • G722
  • G729
  • GSM

Manage your SIP devices with RESTful APIs

SIP device accounts can be created and managed via standard RESTful API calls or manually through the developer portal. Once set up, your SIP devices can authenticate, register, and establish end-to-end sessions and calls. The SIP Server ensures reliable and secure end-to-end connectivity.

We also provide a device monitoring API to track SIP registration and dialog state changes through callback events.

Connect and go with SIP Server as a Service!

Accessing SIP Server functionalities has never been easier. Simply connect your devices and applications to the SIP Server as a Service and get started. You can also connect SIP trunks of your choice to access the PSTN network or establish private connections to applications.

Everything runs in the cloud

No need to set up, manage or maintain a SIP infrastructure.

Usage-based pricing

Pay-as-you-go model – only pay for what you use.

Grow as you go

Built for unlimited scale in terms of size, users, and geography.

Want to start using the SIP Server as a Service? Talk to us today!

Launch your SIP experience!

Get in contact and talk to us!