Voice & Video Services

Technical features

Explore the technical features of our voice & video services. Click on each feature to read more.

SIP Device Management

Manage SIP device accounts using APIs or the customer web portal. Each customer is assigned a unique SIP domain.

Provisioning APIs

  • Authorization API
  • Device Management API
  • SIP Logs API platform
  • Statistics API platform
  • Notification Settings API
  • Trunk provisioning API

SIP Registrar​

Acting as a multi-tenant RFC 3261 SIP Registrar and accepts customer SIP device registrations. SIP user accounts and registration state are maintained in a location service.

SIP Device Monitoring

Notifications can be sent using the AWS SNS service or using HTTPS call-backs to your service of choice.

SIP Proxy

Acting as a multi-tenant RFC 3261 SIP Proxy to relay SIP signaling between different SIP user accounts currently registered within the customer SIP domain.

Hosted NAT Traversal

Includes a media relay and SIP logic to make sure that media flows are delivered end-to-end, even when SIP devices are behind the NAT.

SIP devices must support STUN (RFC 5389) and Symmetric RTP (RFC 4961) in order to guarantee successful NAT traversal.

Hosted SBC

The SIP Server could act as a hosted Session Border Controller (SBC) by letting an internal service connect with SIP devices located on the public internet. Two modes are supported:
  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. It would let a device, that connects using SIP/Websocket for signaling and RTP/DTLS for media, to connect with an internal system even though the internal system uses SIP/TCP and RTP/UDP. The following transport protocols could be mixed and matched:
  • SIP/TCP
  • SIP/UDP
  • SIP/TLS
  • SIP/Websocket
  • RTP/UDP
  • SRTP/UDP
  • RTP/DTLS

Media Types Supported

  • Audio
  • Video

Audio Codecs Supported

For pure proxy service there is no constraints on supported media. For SIP-WebRTC calls the following voice codecs are supported:

  • PCMA
  • PCMU
  • OPUS

Video Codecs Supported

For pure proxy service there is no constraints on supported media. For SIP-WebRTC calls the following video codecs are supported:

  • VP8
  • VP9
  • H.264

SIP Authentication

Authentication of all SIP requests (REGISTER, INVITE et al) using SIP Digest authentication. No user passwords are stored in clear text, only the HA1 is stored.

As an alternative, SIP requests could use a JSON Web token provided in a header for authentication.

SIP Encryption

SIP signaling is encrypted using TLSv1.2 or later. SIP devices must support RFC 5626 and are responsible for keeping the TLS session established to allow inbound SIP signaling.

RTP / Media Encryption

Since media is supposed to flow end-to-end between SIP devices, the SIP Server will not be part of the media flow at all in some cases. When media anchoring is required (e.g. to apply Hosted NAT traversal) the media streams are relayed without any intervention and media remains encrypted end-to-end.

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).

WebRTC Gateway

Supporting SIP over Websocket (WebRTC) and can interwork audio and video sessions between WebRTC and native SIP clients.

Audio Transcoding

Codecs supported:

  • PCMA
  • PCMU
  • OPUS
  • G722
  • G729
  • GSM
5/5
Our customers experience reduced cost, improved time to market and increased quality
iotcomms.io provides state-of-the-art cloud-based services that allow us to connect devices supported by the Navigil Service in a scalable and resilient way to mobile networks.
Jari Huuskonen
Jari HuuskonenManager of Services, Navigil
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