How to configure your firewall and router

Firewall/Router configuration

Usually if you configure your SIP phones clients like recommended in this article Connecting-Phones-via-SIP then no changes need to be made to your router and firewall. The client will look for its own way to pass your NAT and Firewall settings. 

However in some cases it would be necessary to make changes to your firewall and router settings. This would be the case if you encounter problems with sip clients not able to register with our sip proxy or calls getting dropped or if you can't hear audio on one or both sides. If this is the case please change the following settings in your firewall or router.

Allow incoming traffic from babelforce networks

Allow all traffic coming from babelforce servers to pass through your firewall. Depending on in which region you have opened your babelforce account whitelist the following domains and IP addresses:

EU region:

Service Protocol Port Domain / IP address
SIP TCP/UDP  5060 sip.babelforce.com
109.235.234.67
RTP UDP 10000-40000 109.235.232.0/21 *
SIP TCP/UDP 5060 sip.eu.babelforce.com
52.58.110.206
RTP UDP 10000-40000 52.58.110.206
SIP TCP  443 webrtc.babelforce.com
18.185.102.121
 RTP  UDP 10000-40000  18.185.102.121

US region:

Service Protocol Port Domain / IP address
SIP TCP/UDP  5060 sip.us-east.babelforce.com
52.55.242.176
RTP  UDP 10000-40000  52.55.242.176

AP region:

Service Protocol Port IP address
SIP TCP/UDP  5060 sip.ap-southeast.babelforce.com
13.54.226.3
RTP UDP  10000-40000 13.54.226.3

* Please note, this RTP IP address describes a whole network. We make use of different media gateways in our EU region. You need to allow audio traffic coming from all IPs in this network.

UDP / NAT / Advanced SIP settings

Not all firewalls do support configuration of the following features. If your firewall allows to change these settings, we highly recommend it!

  • set UDP-NAT or UDP connection timeout to 180 sec (sometimes this is referred to as UDP inactivity / session / port timeout) 
  • turn on consistent NAT if possible
  • disable any kind of SIP transformations, like for example: SIP ALG (Application Layer Gateway), SIP SPI (Stateful Packet Inspection) and SIP helper or transformations

You can find a high-level description of how network traversal works for SIP here: How to set up your network for babelforce VoIP and SIP

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