This is a set of basic networking requirements and terminologies for connecting to SIPTRUNK.com
SIP.TRUNK has two SIP Gateways which you may connect to:
- Both gateways will only accept SIP traffic on UDP port 5060
Audio related to SIP calls is delivered via RTP over UDP. Since our SIP gateways are just a proxy, the audio can be delivered from various IP addresses and many different ports.
NAT stands for Network Address Translation. Most customers are behind some form of NAT, and many do not have a static IP address. Try to connect your PBX or softphone through our service without any special NAT configuration. If it becomes apparent that you may be having a NAT issue, there are three possible solutions:
STUN: if your PBX or softphone supports it, you may connect to our STUN server at stun.siptrunk.com on port 3478
ALG: If your router or firewall is capable of properly implementing ALG, enabling it may alleviate your issue
STATIC IP: your internet service provider (ISP) may provide a static IP at an additional cost
Network security can be a consideration when implementing a VoIP solution. Here are some basic tips:
1. You should allow incoming SIP traffic from:
- Individual IPs 188.8.131.52 and 184.108.40.206 and Subnets 220.127.116.11/24 and 18.104.22.168/24
- If you cannot allow subnets, then allow individual IPs 22.214.171.124, 126.96.36.199, 188.8.131.52 and 184.108.40.206
2. You should forward all RTP ports used by your device to the private IP address of your device if it is behind NAT
3. Your SIP device should only accept RTP traffic for a SIP call which is active, so the forwarding in tip 2, above, should not be accompanied with blocking traffic from certain IP addresses (see here for an extended explanation).
4. If your device is accessible from the internet, choose a strong password, or preferably, consider disallowing anyone from logging in through the internet