"The game isn’t just about playing — it’s about watching, analyzing, and winning."
In the last level of SIP GAMES, we learned about RTP — the courier that runs your media across the network. But what happens when the game lags, drops, or mysteriously fails?
You don’t guess. You spy.
Not on people, but on packets.
Here’s your VoIP agent toolkit — categorized, explained, and ready for action.
🛡 1. Surveillance Tools — Capture & Analyze Traffic
When something goes wrong in a VoIP call, your first job is to see the traffic. These tools let you peek inside packets like a movie detective flipping through secret files.
🐬 Wireshark
- Purpose: Capture and inspect SIP, SDP, RTP (and more) packets.
- Use case: Call drops after 30 seconds? See if a BYE or timeout happened. No audio? Check if RTP is flowing.
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How to use:
- Install from wireshark.org
- Start capture on the correct network interface.
- Apply filters like:
-
sip→ Show only SIP messages -
rtp→ Show only RTP packets -
ip.addr == 192.168.1.50→ Show traffic for a specific IP
-
- Right-click → “Follow” → “UDP Stream” for audio flow.
- Pro tip: Enable VoIP Calls under Telephony → VoIP Calls to reconstruct call flow.
🧪 2. Simulation Tools — Generate Test Calls
Sometimes you can’t wait for a bug to happen — you need to force it. That’s where call simulators come in.
📞 sipp
- Purpose: SIP traffic generator and test tool.
- Use case: Stress-test your SIP server with thousands of calls, or simulate specific SIP scenarios (registration floods, custom headers, delayed responses).
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How to use:
- Install from sipp.sourceforge.net
- Basic call test:
sipp -sn uac 192.168.1.20(This sends an INVITE to
192.168.1.20using the default UAC scenario.)- Use XML scenario files to define custom call flows.
Pro tip: Combine with Wireshark to capture the generated traffic for later analysis.
🔀 3. Media Manipulation Tools — Intercept & Modify RTP
When you need to play with the media path — for debugging, transcoding, or rerouting — these tools take control.
🎯 rtpengine
- Purpose: RTP proxy for relaying and modifying RTP streams.
- Use case: NAT traversal issues, transcoding between codecs, or enabling SRTP on one side while keeping RTP on the other.
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How to use:
- Install from rtpengine.org
- Use via CLI or integrate with SIP proxies like Kamailio/OpenSIPS.
- Control with commands like:
ngcp-rtpengine-ctl list(Lists active RTP sessions.)
Pro tip: Use it to simulate one-way audio problems by blocking one RTP direction.
🛠 Example Debugging Scenarios
| Scenario | Tool | How It Helps |
|---|---|---|
| One-way audio | Wireshark, rtpengine | Capture RTP to see which side isn’t sending |
| Call setup delay | Wireshark, sipp | Compare SIP INVITE timestamps with responses |
| Codec mismatch | Wireshark | Inspect SDP offers/answers for supported codecs |
| NAT traversal failure | Wireshark, rtpengine | Verify if private IPs appear in SDP |
| Stress testing | sipp | Generate high call volume to see when servers fail |
🎮 TL;DR
- Wireshark = See everything
- sipp = Simulate anything
- rtpengine = Control media flow
- Debugging VoIP isn’t about guessing — it’s about capturing, analyzing, and proving where the problem lives.
🧠 Up Next in SIP GAMES:
“Keeping It Secret: VoIP Security 101” 🔒
We’ll explore SIPS, SRTP, SIP over TLS, RTP over DTLS, and SDES — because in VoIP, sometimes the game isn’t just about speed, but keeping your cards hidden.
Follow @sip_games and stay in the game.
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