SIP Trunking
Connect your PBX directly to our network for seamless voice communication with unlimited channels.
What is SIP Trunking?
SIP (Session Initiation Protocol) trunking is a method of sending voice and other unified communications services over the internet. It replaces traditional telephone lines (PRI/ISDN) by connecting your IP-PBX directly to the public switched telephone network (PSTN) via the internet.
With Telcomet's SIP trunking services, you can consolidate your voice and data communications onto a single network, reducing costs and complexity. Our elastic SIP trunks scale automatically based on your call volume - no need to provision individual channels.
We support all major IP-PBX platforms including Asterisk, FreePBX, 3CX, Cisco, Avaya, and many more. Our technical team can assist with configuration and testing to ensure seamless integration.
Key Benefits
- Unlimited concurrent calls (elastic trunking)
- Compatible with all major PBX systems
- Cost savings vs traditional PRI lines
- HD voice quality with G.722 codec
- Failover and redundancy options
- Number porting support
SIP Trunk Integration
How your PBX connects to the PSTN through our SIP trunk service.
Technical Specifications
Requirements and supported configurations for SIP trunk integration.
Supported Protocols
- SIP over UDP/TCP/TLS
- SRTP for media encryption
- T.38 for fax over IP
- DTMF: RFC2833, SIP INFO, In-band
Supported Codecs
- G.711 (a-law / u-law)
- G.722 (HD Voice)
- G.729
- Opus (upon request)
Compatible PBX Systems
- Asterisk / FreePBX
- 3CX Phone System
- Cisco CUCM / Webex Calling
- Avaya, Mitel, NEC, and more
Network Requirements
- Static IP address (recommended)
- 100 Kbps per concurrent call
- QoS configuration recommended
- Firewall rules for SIP/RTP
Get Started with SIP Trunking
Connect your PBX to our network and start making calls in minutes.