Resilient, reliable, high-quality inbound and outbound voice calls (VoIP) with the most competitive call rates in the market using only global tier one voice carriers.
We provide our partners with SIP trunks that are designed to work successfully with all the leading IP-PBX systems.
Telcom offer a gateway enabling our partners to offer SIP trunking solutions on both IP and Legacy PBX providing their clients with immediate benefit of a flexible and resilient phone service.
Our SIP trunk solutions rates are extremely competitive with very best value in local, national and international voice minutes while maintaining business quality by using global tier one carriers.
- competitive call rates
- Quick to install
- ON- NET SIP solutions
- No packet loss or jitter
- Secure network
- Voice support and escalation process
- Highly-skilled voice engineers support
- Full number porting
- Provision of geo and non-geo number blocks
Our SIP trunk solutions are highly scalable with base call build up to thousands of concurrent calls if required without the expense or lead-time of ordering new lines. We can build a SIP trunk over any IP connection, allowing flexibility on the access technology used.
By using our On-net solutions when provisioned across our own connectivity products your voice or data will not go out over the public internet
Our offering of VoIP service within a dedicated voice VPN (virtual private network) as an optional security feature which allows our partners to provide whatever solution is best suited to your client.
The voice VPN will be protected from all external IP sources. There will be no internet routing within this VPN and the only routes allowed within the voice VRF (virtual routing and forwarding) table are the end customer sites (IP address of the PBX) and the address of the SBC.
QOS – (Quality of service) Our partners can offer voice traffic (VoIP) over access connections and the core network, ensuring voice traffic will always get sent before data streams. We will mark voice packets as EF (expedited forwarding) and with this marking will get priority across our core network, ensuring the low latency necessary for quality voice calls. This will ensure no packet loss or jitter