From the course: Troubleshooting VOIP: Identify and Resolve Common VOIP Issues
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Navigating NAT settings for VoIP
From the course: Troubleshooting VOIP: Identify and Resolve Common VOIP Issues
Navigating NAT settings for VoIP
- [Instructor] When using voice over IP, you may encounter issues because VoIP traffic must pass through a Network Address Translation device. In this segment, we'll outline how NAT configurations such as STUN, TURN, and ICE help prevent issues such as setup failures, one-way audio, or no audio. NAT dynamically translates a private IP address to a public IP address and vice versa. When deploying VoIP in networks that are using Network Address Translation, issues that can affect call quality are common due to how NAT handles traffic. Let's discuss why this happens. VoIP relies on SIP and RTP, which are not designed to handle NAT. SIP will set up the call but does not carry the actual media. RTP handles the media stream based on the connection details provided in the SIP SDP payload. Problems can occur because VoIP packets contain an IP address in the SIP header payload, which NAT does not translate. Let's take a look. We're going to take a look at a couple of SIP headers in Wireshark…
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