ATTENTION: We have transitioned to our enhanced network infrastructure and now utilize gw4.sip.us and gw5.sip.us for all SIP traffic. You must ensure your access rules and phone systems are configured accordingly. We recommend using DNS SRV to ensure redundancy.
If your phone system is only configured for the legacy infrastructure (gw1.sip.us and gw2.sip.us), it will no longer register or receive inbound calls.
This is a set of basic networking requirements and terminologies for connecting to SIP.US.
Gateways
The most efficient way to ensure redundancy is to use DNS SRV. If your PBX supports DNS SRV, pointing to gw.sip.us as your proxy IP address is all that should be necessary to ensure redundancy.
If your PBX does not support DNS SRV, hopefully it supports configuration of multiple outbound proxies. If the system supports a primary and backup SIP proxies, use gw4.sip.us and gw5.sip.us for the backup SIP proxies.
Our gateways will only accept SIP traffic on UDP port 5060.
gw4.sip.us (104.219.162.40)
gw5.sip.us (104.219.163.117)
If you experience issues with resolving our gateways, we recommend trying OpenDNS: 208.67.222.222 & 208.67.220.220.
Audio
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
Most customers are behind some form of Network Address Translation (NAT), and many do not have static IP addresses. 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.sip.us 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.
Security
Network security can be a consideration when implementing a VoIP solution. Here are some basic guidelines:
1. You should allow incoming SIP traffic from:
- Subnets:
- 104.219.162.0/24
- 104.219.163.0/24
- If you cannot allow subnets, allow individual IPs:
- 104.219.162.40
- 104.219.163.117
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.
Fax Server
For Audiocodes SimpleFax ATA service with SIP.US.
Fax Server IP Ranges to Whitelist:
70.97.122.96 – 70.97.122.127 - 70.97.122.96/27
199.242.63.144 – 199.242.63.159 - 199.242.63.144/28
23.175.64.32 – 23.175.64.47 - 23.175.64.32/28
Additionally, regarding ATA fax service, ports 443 and 25 will need to be opened on your firewall. Specifically, make sure nothing is blocking the 443 port.
Please open a ticket with our Support team at support@sip.us if you experience difficulties and need assistance with troubleshooting!