In VoIP Phones

Fixing One Way VoIP Audio (SIP, NAT and STUN)

One of the best inventions of the Internet age has been Voice over IP (VoIP) or in laymen’s terms, the Internet telephone.

This device has ushered in a new era of fast and inexpensive communication for millions of people around the world. However, as with any new technology, there are still bugs that need to be ironed out and one common issue is to place a call only to be rewarded with…nothing (or a one-way conversation which makes making dinner plans tough).

Luckily, fixing this issue is usually straight forward. Usually, the problem is a configuration issue either with the device, the service itself or the network environment.  

Troubleshoot the issue by doing the following:

Update All Firmware

Update the firmware in your routers, the VoIP phones, and any network firewalls.

Update All Software

Make sure all of your devices are running the latest version of software. Make sure the software is configured correctly to allow VoIP communications. Often times, this is simply a matter of checking the appropriate checkbox on a configuration page.

NAT Configuration

Network Address Translation (NAT) is a commonly used in many networks, but unfortunately can cause issues with VoIP. NAT acts as an IP address mask and can prevent VoIP devices from establishing the connections needed for a voice call.

If applicable, either you or someone on your IT team can use a network analysis tool to identify and capture SIP and RTP packets in the call path. A tool such as WireShark can help identify where the issues might be arising.

For example:

VOIP phone —–<1>—– network firewall —–<2>—– SIP proxy —–<3>——network firewall —–<4>—>

  1. Start capturing at point 1
  2. Make a VOIP call.
  3. Analyze capture and identify issue.
  4. If no issue is identified, go to step 2, 3 and so forth.
  5. Fix identified issues and retest.

 

Simple Traversal of UDP through NAT (STUN)

Using STUN allows your VoIP phone system to navigate around the various obstacles thrown up by NAT. STUN is an industry standard and the technical details are published in RFC 3489.

Your phone will require access to a STUN server (most likely provided by your VoIP service provider). However, any phone can use any STUN server, so if your provider cannot provide you an IP address, don’t fret (stun.xten.com, stun.sipgate.net, stunserver.org are all address you can try).

Consult your device manual to locate exactly where this IP address or URL should be entered and reboot the phone.

Note: not all VoIP phones are STUN compatible.

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