Troubleshooting One-Way Audio in SIP: A Wireshark Guide

The Dreaded "Hello? Can You Hear Me?"

We've all been there. The call aligns, the phones ring, but when someone speaks, there's just... silence. Or worse, one person is talking into the void. One-way audio is the most common and frustrating issue in VoIP. It’s almost always a network issue, but proving it is the hard part.

In this guide, I'll walk you through exactly how I diagnose this using Wireshark. No fluff, just the steps.

Step 1: Capture the Traffic

You can't fix what you can't see. Start a packet capture on your media server or gateway. If you're using Linux:

tcpdump -i eth0 -s 0 -w oneway_audio.pcap port 5060 or portrange 10000-20000

Make sure to capture both the signaling (SIP) and the media (RTP).

Step 2: Analyze the SDP (Session Description Protocol)

The SIP INVITE and 200 OK messages contain the SDP, which tells the phones where to send audio. Open the PCAP in Wireshark and filter for sip.

Look specifically at:

• c=IN IP4 x.x.x.x: Is this IP reachable? If it's a private IP (192.168.x.x) and you're traversing the internet, the other side won't be able to send audio to it. This is a classic NAT issue.
• m=audio 12345 RTP/AVP 0: Is the port open on your firewall?

Step 3: The "RTP Streams" Feature

This is Wireshark's superpower. Go to Telephony > RTP > RTP Streams.

You should see two streams for a healthy call: one forward, one reverse. If you only see one, or if one has a packet count of 0, the audio isn't even reaching your capture point.

Conclusion

Fixing one-way audio usually involves configuring your SIP server (like FreeSWITCH or Asterisk) to advertise its Public IP in the SDP, or setting up a STUN/TURN server to handle the NAT traversal.

References

• RFC 4566 - SDP: Session Description Protocol
• Wireshark User Guide - RTP Analysis